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

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* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (367 commits)
  ALSA: ASoC: fix a typo in omp-pcm.c
  ASoC: Fix DSP formats in SSM2602 audio codec
  ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers
  ALSA: hda: fix incorrect mixer index values for 92hd83xx
  ALSA: hda: dinput_mux check
  ALSA: hda - Add quirk for another HP dv7
  ALSA: ASoC - Add missing __devexit annotation to wm8350.c
  ALSA: ASoc: DaVinci: davinci-evm use dsp_b mode
  ALSA: ASoC: DaVinci: i2s, evm, pass same value to codec and cpu_dai
  ALSA: ASoC: tlv320aic3x add dsp_a
  ALSA: ASoC: DaVinci: document I2S limitations
  ALSA: ASoC: DaVinci: davinci-i2s clean up
  ALSA: ASoC: DaVinci: davinci-i2s clean up
  ALSA: ASoC: DaVinci: davinci-i2s add comments to explain polarity
  ALSA: ASoC: DaVinci: davinvi-evm, make requests explicit
  ALSA: ca0106 - disable 44.1kHz capture
  ALSA: ca0106 - Add missing card->private_data initialization
  ALSA: ca0106 - Check ac97 availability at PM
  ALSA: hda - Power up always when no jack detection is available
  ALSA: hda - Fix unused variable warnings in patch_sigmatel.c
  ...
This commit is contained in:
Linus Torvalds 2008-12-28 11:41:32 -08:00
commit cb10ea549f
253 changed files with 22171 additions and 8087 deletions
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330
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@ -757,6 +757,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
model - force the model name
position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF)
probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
probe_only - Only probing and no codec initialization (default=off);
Useful to check the initial codec status for debugging
bdl_pos_adj - Specifies the DMA IRQ timing delay in samples.
Passing -1 will make the driver to choose the appropriate
value based on the controller chip.
@ -772,327 +774,23 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
This module supports multiple cards and autoprobe.
See Documentation/sound/alsa/HD-Audio.txt for more details about
HD-audio driver.
Each codec may have a model table for different configurations.
If your machine isn't listed there, the default (usually minimal)
configuration is set up. You can pass "model=<name>" option to
specify a certain model in such a case. There are different
models depending on the codec chip.
Model name Description
---------- -----------
ALC880
3stack 3-jack in back and a headphone out
3stack-digout 3-jack in back, a HP out and a SPDIF out
5stack 5-jack in back, 2-jack in front
5stack-digout 5-jack in back, 2-jack in front, a SPDIF out
6stack 6-jack in back, 2-jack in front
6stack-digout 6-jack with a SPDIF out
w810 3-jack
z71v 3-jack (HP shared SPDIF)
asus 3-jack (ASUS Mobo)
asus-w1v ASUS W1V
asus-dig ASUS with SPDIF out
asus-dig2 ASUS with SPDIF out (using GPIO2)
uniwill 3-jack
fujitsu Fujitsu Laptops (Pi1536)
F1734 2-jack
lg LG laptop (m1 express dual)
lg-lw LG LW20/LW25 laptop
tcl TCL S700
clevo Clevo laptops (m520G, m665n)
medion Medion Rim 2150
test for testing/debugging purpose, almost all controls can be
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC260
hp HP machines
hp-3013 HP machines (3013-variant)
hp-dc7600 HP DC7600
fujitsu Fujitsu S7020
acer Acer TravelMate
will Will laptops (PB V7900)
replacer Replacer 672V
basic fixed pin assignment (old default model)
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC262
fujitsu Fujitsu Laptop
hp-bpc HP xw4400/6400/8400/9400 laptops
hp-bpc-d7000 HP BPC D7000
hp-tc-t5735 HP Thin Client T5735
hp-rp5700 HP RP5700
benq Benq ED8
benq-t31 Benq T31
hippo Hippo (ATI) with jack detection, Sony UX-90s
hippo_1 Hippo (Benq) with jack detection
sony-assamd Sony ASSAMD
toshiba-s06 Toshiba S06
toshiba-rx1 Toshiba RX1
ultra Samsung Q1 Ultra Vista model
lenovo-3000 Lenovo 3000 y410
nec NEC Versa S9100
basic fixed pin assignment w/o SPDIF
auto auto-config reading BIOS (default)
ALC267/268
quanta-il1 Quanta IL1 mini-notebook
3stack 3-stack model
toshiba Toshiba A205
acer Acer laptops
acer-aspire Acer Aspire One
dell Dell OEM laptops (Vostro 1200)
zepto Zepto laptops
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC269
basic Basic preset
quanta Quanta FL1
eeepc-p703 ASUS Eeepc P703 P900A
eeepc-p901 ASUS Eeepc P901 S101
ALC662/663
3stack-dig 3-stack (2-channel) with SPDIF
3stack-6ch 3-stack (6-channel)
3stack-6ch-dig 3-stack (6-channel) with SPDIF
6stack-dig 6-stack with SPDIF
lenovo-101e Lenovo laptop
eeepc-p701 ASUS Eeepc P701
eeepc-ep20 ASUS Eeepc EP20
ecs ECS/Foxconn mobo
m51va ASUS M51VA
g71v ASUS G71V
h13 ASUS H13
g50v ASUS G50V
asus-mode1 ASUS
asus-mode2 ASUS
asus-mode3 ASUS
asus-mode4 ASUS
asus-mode5 ASUS
asus-mode6 ASUS
auto auto-config reading BIOS (default)
ALC882/885
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
arima Arima W820Di1
targa Targa T8, MSI-1049 T8
asus-a7j ASUS A7J
asus-a7m ASUS A7M
macpro MacPro support
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
w2jc ASUS W2JC
auto auto-config reading BIOS (default)
ALC883/888
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
3stack-6ch 3-jack 6-channel
3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
6stack-dig-demo 6-jack digital for Intel demo board
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
acer-aspire Acer Aspire 9810
medion Medion Laptops
medion-md2 Medion MD2
targa-dig Targa/MSI
targa-2ch-dig Targs/MSI with 2-channel
laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
lenovo-101e Lenovo 101E
lenovo-nb0763 Lenovo NB0763
lenovo-ms7195-dig Lenovo MS7195
lenovo-sky Lenovo Sky
haier-w66 Haier W66
3stack-hp HP machines with 3stack (Lucknow, Samba boards)
6stack-dell Dell machines with 6stack (Inspiron 530)
mitac Mitac 8252D
clevo-m720 Clevo M720 laptop series
fujitsu-pi2515 Fujitsu AMILO Pi2515
3stack-6ch-intel Intel DG33* boards
auto auto-config reading BIOS (default)
ALC861/660
3stack 3-jack
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack with SPDIF I/O
3stack-660 3-jack (for ALC660)
uniwill-m31 Uniwill M31 laptop
toshiba Toshiba laptop support
asus Asus laptop support
asus-laptop ASUS F2/F3 laptops
auto auto-config reading BIOS (default)
ALC861VD/660VD
3stack 3-jack
3stack-dig 3-jack with SPDIF OUT
6stack-dig 6-jack with SPDIF OUT
3stack-660 3-jack (for ALC660VD)
3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
lenovo Lenovo 3000 C200
dallas Dallas laptops
hp HP TX1000
auto auto-config reading BIOS (default)
CMI9880
minimal 3-jack in back
min_fp 3-jack in back, 2-jack in front
full 6-jack in back, 2-jack in front
full_dig 6-jack in back, 2-jack in front, SPDIF I/O
allout 5-jack in back, 2-jack in front, SPDIF out
auto auto-config reading BIOS (default)
AD1882 / AD1882A
3stack 3-stack mode (default)
6stack 6-stack mode
AD1884A / AD1883 / AD1984A / AD1984B
desktop 3-stack desktop (default)
laptop laptop with HP jack sensing
mobile mobile devices with HP jack sensing
thinkpad Lenovo Thinkpad X300
AD1884
N/A
AD1981
basic 3-jack (default)
hp HP nx6320
thinkpad Lenovo Thinkpad T60/X60/Z60
toshiba Toshiba U205
AD1983
N/A
AD1984
basic default configuration
thinkpad Lenovo Thinkpad T61/X61
dell Dell T3400
AD1986A
6stack 6-jack, separate surrounds (default)
3stack 3-stack, shared surrounds
laptop 2-channel only (FSC V2060, Samsung M50)
laptop-eapd 2-channel with EAPD (Samsung R65, ASUS A6J)
laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
ultra 2-channel with EAPD (Samsung Ultra tablet PC)
AD1988/AD1988B/AD1989A/AD1989B
6stack 6-jack
6stack-dig ditto with SPDIF
3stack 3-jack
3stack-dig ditto with SPDIF
laptop 3-jack with hp-jack automute
laptop-dig ditto with SPDIF
auto auto-config reading BIOS (default)
Conexant 5045
laptop-hpsense Laptop with HP sense (old model laptop)
laptop-micsense Laptop with Mic sense (old model fujitsu)
laptop-hpmicsense Laptop with HP and Mic senses
benq Benq R55E
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
Conexant 5047
laptop Basic Laptop config
laptop-hp Laptop config for some HP models (subdevice 30A5)
laptop-eapd Laptop config with EAPD support
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
Conexant 5051
laptop Basic Laptop config (default)
hp HP Spartan laptop
STAC9200
ref Reference board
dell-d21 Dell (unknown)
dell-d22 Dell (unknown)
dell-d23 Dell (unknown)
dell-m21 Dell Inspiron 630m, Dell Inspiron 640m
dell-m22 Dell Latitude D620, Dell Latitude D820
dell-m23 Dell XPS M1710, Dell Precision M90
dell-m24 Dell Latitude 120L
dell-m25 Dell Inspiron E1505n
dell-m26 Dell Inspiron 1501
dell-m27 Dell Inspiron E1705/9400
gateway Gateway laptops with EAPD control
panasonic Panasonic CF-74
STAC9205/9254
ref Reference board
dell-m42 Dell (unknown)
dell-m43 Dell Precision
dell-m44 Dell Inspiron
STAC9220/9221
ref Reference board
3stack D945 3stack
5stack D945 5stack + SPDIF
intel-mac-v1 Intel Mac Type 1
intel-mac-v2 Intel Mac Type 2
intel-mac-v3 Intel Mac Type 3
intel-mac-v4 Intel Mac Type 4
intel-mac-v5 Intel Mac Type 5
intel-mac-auto Intel Mac (detect type according to subsystem id)
macmini Intel Mac Mini (equivalent with type 3)
macbook Intel Mac Book (eq. type 5)
macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3)
imac-intel Intel iMac (eq. type 2)
imac-intel-20 Intel iMac (newer version) (eq. type 3)
dell-d81 Dell (unknown)
dell-d82 Dell (unknown)
dell-m81 Dell (unknown)
dell-m82 Dell XPS M1210
STAC9202/9250/9251
ref Reference board, base config
m2-2 Some Gateway MX series laptops
m6 Some Gateway NX series laptops
pa6 Gateway NX860 series
STAC9227/9228/9229/927x
ref Reference board
ref-no-jd Reference board without HP/Mic jack detection
3stack D965 3stack
5stack D965 5stack + SPDIF
dell-3stack Dell Dimension E520
dell-bios Fixes with Dell BIOS setup
STAC92HD71B*
ref Reference board
dell-m4-1 Dell desktops
dell-m4-2 Dell desktops
dell-m4-3 Dell desktops
STAC92HD73*
ref Reference board
no-jd BIOS setup but without jack-detection
dell-m6-amic Dell desktops/laptops with analog mics
dell-m6-dmic Dell desktops/laptops with digital mics
dell-m6 Dell desktops/laptops with both type of mics
STAC9872
vaio Setup for VAIO FE550G/SZ110
vaio-ar Setup for VAIO AR
models depending on the codec chip. The list of available models
is found in HD-Audio-Models.txt
The model name "genric" is treated as a special case. When this
model is given, the driver uses the generic codec parser without
"codec-patch". It's sometimes good for testing and debugging.
If the default configuration doesn't work and one of the above
matches with your device, report it together with the PCI
subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
matches with your device, report it together with alsa-info.sh
output (with --no-upload option) to kernel bugzilla or alsa-devel
ML (see the section "Links and Addresses").
power_save and power_save_controller options are for power-saving
@ -1652,7 +1350,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
* AuzenTech X-Meridian
* Bgears b-Enspirer
* Club3D Theatron DTS
* HT-Omega Claro
* HT-Omega Claro (plus)
* HT-Omega Claro halo (XT)
* Razer Barracuda AC-1
* Sondigo Inferno
@ -2409,8 +2108,11 @@ Links and Addresses
ALSA project homepage
http://www.alsa-project.org
ALSA Bug Tracking System
https://bugtrack.alsa-project.org/bugs/
Kernel Bugzilla
http://bugzilla.kernel.org/
ALSA Developers ML
mailto:alsa-devel@alsa-project.org
alsa-info.sh script
http://www.alsa-project.org/alsa-info.sh

348
Documentation/sound/alsa/HD-Audio-Models.txt Normal file
View File

@ -0,0 +1,348 @@
Model name Description
---------- -----------
ALC880
======
3stack 3-jack in back and a headphone out
3stack-digout 3-jack in back, a HP out and a SPDIF out
5stack 5-jack in back, 2-jack in front
5stack-digout 5-jack in back, 2-jack in front, a SPDIF out
6stack 6-jack in back, 2-jack in front
6stack-digout 6-jack with a SPDIF out
w810 3-jack
z71v 3-jack (HP shared SPDIF)
asus 3-jack (ASUS Mobo)
asus-w1v ASUS W1V
asus-dig ASUS with SPDIF out
asus-dig2 ASUS with SPDIF out (using GPIO2)
uniwill 3-jack
fujitsu Fujitsu Laptops (Pi1536)
F1734 2-jack
lg LG laptop (m1 express dual)
lg-lw LG LW20/LW25 laptop
tcl TCL S700
clevo Clevo laptops (m520G, m665n)
medion Medion Rim 2150
test for testing/debugging purpose, almost all controls can be
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC260
======
hp HP machines
hp-3013 HP machines (3013-variant)
hp-dc7600 HP DC7600
fujitsu Fujitsu S7020
acer Acer TravelMate
will Will laptops (PB V7900)
replacer Replacer 672V
basic fixed pin assignment (old default model)
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC262
======
fujitsu Fujitsu Laptop
hp-bpc HP xw4400/6400/8400/9400 laptops
hp-bpc-d7000 HP BPC D7000
hp-tc-t5735 HP Thin Client T5735
hp-rp5700 HP RP5700
benq Benq ED8
benq-t31 Benq T31
hippo Hippo (ATI) with jack detection, Sony UX-90s
hippo_1 Hippo (Benq) with jack detection
sony-assamd Sony ASSAMD
toshiba-s06 Toshiba S06
toshiba-rx1 Toshiba RX1
ultra Samsung Q1 Ultra Vista model
lenovo-3000 Lenovo 3000 y410
nec NEC Versa S9100
basic fixed pin assignment w/o SPDIF
auto auto-config reading BIOS (default)
ALC267/268
==========
quanta-il1 Quanta IL1 mini-notebook
3stack 3-stack model
toshiba Toshiba A205
acer Acer laptops
acer-dmic Acer laptops with digital-mic
acer-aspire Acer Aspire One
dell Dell OEM laptops (Vostro 1200)
zepto Zepto laptops
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC269
======
basic Basic preset
quanta Quanta FL1
eeepc-p703 ASUS Eeepc P703 P900A
eeepc-p901 ASUS Eeepc P901 S101
fujitsu FSC Amilo
auto auto-config reading BIOS (default)
ALC662/663
==========
3stack-dig 3-stack (2-channel) with SPDIF
3stack-6ch 3-stack (6-channel)
3stack-6ch-dig 3-stack (6-channel) with SPDIF
6stack-dig 6-stack with SPDIF
lenovo-101e Lenovo laptop
eeepc-p701 ASUS Eeepc P701
eeepc-ep20 ASUS Eeepc EP20
ecs ECS/Foxconn mobo
m51va ASUS M51VA
g71v ASUS G71V
h13 ASUS H13
g50v ASUS G50V
asus-mode1 ASUS
asus-mode2 ASUS
asus-mode3 ASUS
asus-mode4 ASUS
asus-mode5 ASUS
asus-mode6 ASUS
auto auto-config reading BIOS (default)
ALC882/885
==========
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
arima Arima W820Di1
targa Targa T8, MSI-1049 T8
asus-a7j ASUS A7J
asus-a7m ASUS A7M
macpro MacPro support
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
w2jc ASUS W2JC
auto auto-config reading BIOS (default)
ALC883/888
==========
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
3stack-6ch 3-jack 6-channel
3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
6stack-dig-demo 6-jack digital for Intel demo board
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
acer-aspire Acer Aspire 9810
acer-aspire-4930g Acer Aspire 4930G
medion Medion Laptops
medion-md2 Medion MD2
targa-dig Targa/MSI
targa-2ch-dig Targs/MSI with 2-channel
laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
lenovo-101e Lenovo 101E
lenovo-nb0763 Lenovo NB0763
lenovo-ms7195-dig Lenovo MS7195
lenovo-sky Lenovo Sky
haier-w66 Haier W66
3stack-hp HP machines with 3stack (Lucknow, Samba boards)
6stack-dell Dell machines with 6stack (Inspiron 530)
mitac Mitac 8252D
clevo-m720 Clevo M720 laptop series
fujitsu-pi2515 Fujitsu AMILO Pi2515
fujitsu-xa3530 Fujitsu AMILO XA3530
3stack-6ch-intel Intel DG33* boards
auto auto-config reading BIOS (default)
ALC861/660
==========
3stack 3-jack
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack with SPDIF I/O
3stack-660 3-jack (for ALC660)
uniwill-m31 Uniwill M31 laptop
toshiba Toshiba laptop support
asus Asus laptop support
asus-laptop ASUS F2/F3 laptops
auto auto-config reading BIOS (default)
ALC861VD/660VD
==============
3stack 3-jack
3stack-dig 3-jack with SPDIF OUT
6stack-dig 6-jack with SPDIF OUT
3stack-660 3-jack (for ALC660VD)
3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
lenovo Lenovo 3000 C200
dallas Dallas laptops
hp HP TX1000
asus-v1s ASUS V1Sn
auto auto-config reading BIOS (default)
CMI9880
=======
minimal 3-jack in back
min_fp 3-jack in back, 2-jack in front
full 6-jack in back, 2-jack in front
full_dig 6-jack in back, 2-jack in front, SPDIF I/O
allout 5-jack in back, 2-jack in front, SPDIF out
auto auto-config reading BIOS (default)
AD1882 / AD1882A
================
3stack 3-stack mode (default)
6stack 6-stack mode
AD1884A / AD1883 / AD1984A / AD1984B
====================================
desktop 3-stack desktop (default)
laptop laptop with HP jack sensing
mobile mobile devices with HP jack sensing
thinkpad Lenovo Thinkpad X300
AD1884
======
N/A
AD1981
======
basic 3-jack (default)
hp HP nx6320
thinkpad Lenovo Thinkpad T60/X60/Z60
toshiba Toshiba U205
AD1983
======
N/A
AD1984
======
basic default configuration
thinkpad Lenovo Thinkpad T61/X61
dell Dell T3400
AD1986A
=======
6stack 6-jack, separate surrounds (default)
3stack 3-stack, shared surrounds
laptop 2-channel only (FSC V2060, Samsung M50)
laptop-eapd 2-channel with EAPD (ASUS A6J)
laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
ultra 2-channel with EAPD (Samsung Ultra tablet PC)
samsung 2-channel with EAPD (Samsung R65)
AD1988/AD1988B/AD1989A/AD1989B
==============================
6stack 6-jack
6stack-dig ditto with SPDIF
3stack 3-jack
3stack-dig ditto with SPDIF
laptop 3-jack with hp-jack automute
laptop-dig ditto with SPDIF
auto auto-config reading BIOS (default)
Conexant 5045
=============
laptop-hpsense Laptop with HP sense (old model laptop)
laptop-micsense Laptop with Mic sense (old model fujitsu)
laptop-hpmicsense Laptop with HP and Mic senses
benq Benq R55E
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
Conexant 5047
=============
laptop Basic Laptop config
laptop-hp Laptop config for some HP models (subdevice 30A5)
laptop-eapd Laptop config with EAPD support
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
Conexant 5051
=============
laptop Basic Laptop config (default)
hp HP Spartan laptop
STAC9200
========
ref Reference board
dell-d21 Dell (unknown)
dell-d22 Dell (unknown)
dell-d23 Dell (unknown)
dell-m21 Dell Inspiron 630m, Dell Inspiron 640m
dell-m22 Dell Latitude D620, Dell Latitude D820
dell-m23 Dell XPS M1710, Dell Precision M90
dell-m24 Dell Latitude 120L
dell-m25 Dell Inspiron E1505n
dell-m26 Dell Inspiron 1501
dell-m27 Dell Inspiron E1705/9400
gateway Gateway laptops with EAPD control
panasonic Panasonic CF-74
STAC9205/9254
=============
ref Reference board
dell-m42 Dell (unknown)
dell-m43 Dell Precision
dell-m44 Dell Inspiron
STAC9220/9221
=============
ref Reference board
3stack D945 3stack
5stack D945 5stack + SPDIF
intel-mac-v1 Intel Mac Type 1
intel-mac-v2 Intel Mac Type 2
intel-mac-v3 Intel Mac Type 3
intel-mac-v4 Intel Mac Type 4
intel-mac-v5 Intel Mac Type 5
intel-mac-auto Intel Mac (detect type according to subsystem id)
macmini Intel Mac Mini (equivalent with type 3)
macbook Intel Mac Book (eq. type 5)
macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3)
imac-intel Intel iMac (eq. type 2)
imac-intel-20 Intel iMac (newer version) (eq. type 3)
dell-d81 Dell (unknown)
dell-d82 Dell (unknown)
dell-m81 Dell (unknown)
dell-m82 Dell XPS M1210
STAC9202/9250/9251
==================
ref Reference board, base config
m2-2 Some Gateway MX series laptops
m6 Some Gateway NX series laptops
pa6 Gateway NX860 series
STAC9227/9228/9229/927x
=======================
ref Reference board
ref-no-jd Reference board without HP/Mic jack detection
3stack D965 3stack
5stack D965 5stack + SPDIF
dell-3stack Dell Dimension E520
dell-bios Fixes with Dell BIOS setup
STAC92HD71B*
============
ref Reference board
dell-m4-1 Dell desktops
dell-m4-2 Dell desktops
dell-m4-3 Dell desktops
STAC92HD73*
===========
ref Reference board
no-jd BIOS setup but without jack-detection
dell-m6-amic Dell desktops/laptops with analog mics
dell-m6-dmic Dell desktops/laptops with digital mics
dell-m6 Dell desktops/laptops with both type of mics
STAC92HD83*
===========
ref Reference board
STAC9872
========
vaio Setup for VAIO FE550G/SZ110
vaio-ar Setup for VAIO AR

577
Documentation/sound/alsa/HD-Audio.txt Normal file
View File

@ -0,0 +1,577 @@
MORE NOTES ON HD-AUDIO DRIVER
=============================
Takashi Iwai <tiwai@suse.de>
GENERAL
-------
HD-audio is the new standard on-board audio component on modern PCs
after AC97. Although Linux has been supporting HD-audio since long
time ago, there are often problems with new machines. A part of the
problem is broken BIOS, and the rest is the driver implementation.
This document explains the brief trouble-shooting and debugging
methods for the HD-audio hardware.
The HD-audio component consists of two parts: the controller chip and
the codec chips on the HD-audio bus. Linux provides a single driver
for all controllers, snd-hda-intel. Although the driver name contains
a word of a well-known harware vendor, it's not specific to it but for
all controller chips by other companies. Since the HD-audio
controllers are supposed to be compatible, the single snd-hda-driver
should work in most cases. But, not surprisingly, there are known
bugs and issues specific to each controller type. The snd-hda-intel
driver has a bunch of workarounds for these as described below.
A controller may have multiple codecs. Usually you have one audio
codec and optionally one modem codec. In theory, there might be
multiple audio codecs, e.g. for analog and digital outputs, and the
driver might not work properly because of conflict of mixer elements.
This should be fixed in future if such hardware really exists.
The snd-hda-intel driver has several different codec parsers depending
on the codec. It has a generic parser as a fallback, but this
functionality is fairly limited until now. Instead of the generic
parser, usually the codec-specific parser (coded in patch_*.c) is used
for the codec-specific implementations. The details about the
codec-specific problems are explained in the later sections.
If you are interested in the deep debugging of HD-audio, read the
HD-audio specification at first. The specification is found on
Intel's web page, for example:
- http://www.intel.com/standards/hdaudio/
HD-AUDIO CONTROLLER
-------------------
DMA-Position Problem
~~~~~~~~~~~~~~~~~~~~
The most common problem of the controller is the inaccurate DMA
pointer reporting. The DMA pointer for playback and capture can be
read in two ways, either via a LPIB register or via a position-buffer
map. As default the driver tries to read from the io-mapped
position-buffer, and falls back to LPIB if the position-buffer appears
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
help.
In addition to that, every controller is known to be broken regarding
the wake-up timing. It wakes up a few samples before actually
processing the data on the buffer. This caused a lot of problems, for
example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts
an artificial delay to the wake up timing. This delay is controlled
via `bdl_pos_adj` option.
When `bdl_pos_adj` is a negative value (as default), it's assigned to
an appropriate value depending on the controller chip. For Intel
chips, it'd be 1 while it'd be 32 for others. Usually this works.
Only in case it doesn't work and you get warning messages, you should
change this parameter to other values.
Codec-Probing Problem
~~~~~~~~~~~~~~~~~~~~~
A less often but a more severe problem is the codec probing. When
BIOS reports the available codec slots wrongly, the driver gets
confused and tries to access the non-existing codec slot. This often
results in the total screw-up, and destructs the further communication
with the codec chips. The symptom appears usually as error messages
like:
------------------------------------------------------------------------
hda_intel: azx_get_response timeout, switching to polling mode:
last cmd=0x12345678
hda_intel: azx_get_response timeout, switching to single_cmd mode:
last cmd=0x12345678
------------------------------------------------------------------------
The first line is a warning, and this is usually relatively harmless.
It means that the codec response isn't notified via an IRQ. The
driver uses explicit polling method to read the response. It gives
very slight CPU overhead, but you'd unlikely notice it.
The second line is, however, a fatal error. If this happens, usually
it means that something is really wrong. Most likely you are
accessing a non-existing codec slot.
Thus, if the second error message appears, try to narrow the probed
codec slots via `probe_mask` option. It's a bitmask, and each bit
corresponds to the codec slot. For example, to probe only the first
slot, pass `probe_mask=1`. For the first and the third slots, pass
`probe_mask=5` (where 5 = 1 | 4), and so on.
Since 2.6.29 kernel, the driver has a more robust probing method, so
this error might happen rarely, though.
Interrupt Handling
~~~~~~~~~~~~~~~~~~
In rare but some cases, the interrupt isn't properly handled as
default. You would notice this by the DMA transfer error reported by
ALSA PCM core, for example. Using MSI might help in such a case.
Pass `enable_msi=1` option for enabling MSI.
HD-AUDIO CODEC
--------------
Model Option
~~~~~~~~~~~~
The most common problem regarding the HD-audio driver is the
unsupported codec features or the mismatched device configuration.
Most of codec-specific code has several preset models, either to
override the BIOS setup or to provide more comprehensive features.
The driver checks PCI SSID and looks through the static configuration
table until any matching entry is found. If you have a new machine,
you may see a message like below:
------------------------------------------------------------------------
hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...
------------------------------------------------------------------------
Even if you see such a message, DON'T PANIC. Take a deep breath and
keep your towel. First of all, it's an informational message, no
warning, no error. This means that the PCI SSID of your device isn't
listed in the known preset model (white-)list. But, this doesn't mean
that the driver is broken. Many codec-drivers provide the automatic
configuration mechanism based on the BIOS setup.
The HD-audio codec has usually "pin" widgets, and BIOS sets the default
configuration of each pin, which indicates the location, the
connection type, the jack color, etc. The HD-audio driver can guess
the right connection judging from these default configuration values.
However -- some codec-support codes, such as patch_analog.c, don't
support the automatic probing (yet as of 2.6.28). And, BIOS is often,
yes, pretty often broken. It sets up wrong values and screws up the
driver.
The preset model is provided basically to overcome such a situation.
When the matching preset model is found in the white-list, the driver
assumes the static configuration of that preset and builds the mixer
elements and PCM streams based on the static information. Thus, if
you have a newer machine with a slightly different PCI SSID from the
existing one, you may have a good chance to re-use the same model.
You can pass the `model` option to specify the preset model instead of
PCI SSID look-up.
What `model` option values are available depends on the codec chip.
Check your codec chip from the codec proc file (see "Codec Proc-File"
section below). It will show the vendor/product name of your codec
chip. Then, see Documentation/sound/alsa/HD-Audio-Modelstxt file,
the section of HD-audio driver. You can find a list of codecs
and `model` options belonging to each codec. For example, for Realtek
ALC262 codec chip, pass `model=ultra` for devices that are compatible
with Samsung Q1 Ultra.
Thus, the first thing you can do for any brand-new, unsupported and
non-working HD-audio hardware is to check HD-audio codec and several
different `model` option values. If you have a luck, some of them
might suit with your device well.
Some codecs such as ALC880 have a special model option `model=test`.
This configures the driver to provide as many mixer controls as
possible for every single pin feature except for the unsolicited
events (and maybe some other specials). Adjust each mixer element and
try the I/O in the way of trial-and-error until figuring out the whole
I/O pin mappings.
Note that `model=generic` has a special meaning. It means to use the
generic parser regardless of the codec. Usually the codec-specific
parser is much better than the generic parser (as now). Thus this
option is more about the debugging purpose.
Speaker and Headphone Output
~~~~~~~~~~~~~~~~~~~~~~~~~~~~
One of the most frequent (and obvious) bugs with HD-audio is the
silent output from either or both of a built-in speaker and a
headphone jack. In general, you should try a headphone output at
first. A speaker output often requires more additional controls like
the external amplifier bits. Thus a headphone output has a slightly
better chance.
Before making a bug report, double-check whether the mixer is set up
correctly. The recent version of snd-hda-intel driver provides mostly
"Master" volume control as well as "Front" volume (where Front
indicates the front-channels). In addition, there can be individual
"Headphone" and "Speaker" controls.
Ditto for the speaker output. There can be "External Amplifier"
switch on some codecs. Turn on this if present.
Another related problem is the automatic mute of speaker output by
headphone plugging. This feature is implemented in most cases, but
not on every preset model or codec-support code.
In anyway, try a different model option if you have such a problem.
Some other models may match better and give you more matching
functionality. If none of the available models works, send a bug
report. See the bug report section for details.
If you are masochistic enough to debug the driver problem, note the
following:
- The speaker (and the headphone, too) output often requires the
external amplifier. This can be set usually via EAPD verb or a
certain GPIO. If the codec pin supports EAPD, you have a better
chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly
it's either GPIO0 or GPIO1) may turn on/off EAPD.
- Some Realtek codecs require special vendor-specific coefficients to
turn on the amplifier. See patch_realtek.c.
- IDT codecs may have extra power-enable/disable controls on each
analog pin. See patch_sigmatel.c.
- Very rare but some devices don't accept the pin-detection verb until
triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the
codec-communication stall. Some examples are found in
patch_realtek.c.
Capture Problems
~~~~~~~~~~~~~~~~
The capture problems are often because of missing setups of mixers.
Thus, before submitting a bug report, make sure that you set up the
mixer correctly. For example, both "Capture Volume" and "Capture
Switch" have to be set properly in addition to the right "Capture
Source" or "Input Source" selection. Some devices have "Mic Boost"
volume or switch.
When the PCM device is opened via "default" PCM (without pulse-audio
plugin), you'll likely have "Digital Capture Volume" control as well.
This is provided for the extra gain/attenuation of the signal in
software, especially for the inputs without the hardware volume
control such as digital microphones. Unless really needed, this
should be set to exactly 50%, corresponding to 0dB -- neither extra
gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM,
this control will have no influence, though.
It's known that some codecs / devices have fairly bad analog circuits,
and the recorded sound contains a certain DC-offset. This is no bug
of the driver.
Most of modern laptops have no analog CD-input connection. Thus, the
recording from CD input won't work in many cases although the driver
provides it as the capture source. Use CDDA instead.
The automatic switching of the built-in and external mic per plugging
is implemented on some codec models but not on every model. Partly
because of my laziness but mostly lack of testers. Feel free to
submit the improvement patch to the author.
Direct Debugging
~~~~~~~~~~~~~~~~
If no model option gives you a better result, and you are a tough guy
to fight against evil, try debugging via hitting the raw HD-audio
codec verbs to the device. Some tools are available: hda-emu and
hda-analyzer. The detailed description is found in the sections
below. You'd need to enable hwdep for using these tools. See "Kernel
Configuration" section.
OTHER ISSUES
------------
Kernel Configuration
~~~~~~~~~~~~~~~~~~~~
In general, I recommend you to enable the sound debug option,
`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not.
This enables snd_printd() macro and others, and you'll get additional
kernel messages at probing.
In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`. But this
will give you far more messages. Thus turn this on only when you are
sure to want it.
Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*`
options. Note that each of them corresponds to the codec chip, not
the controller chip. Thus, even if lspci shows the Nvidia controller,
you may need to choose the option for other vendors. If you are
unsure, just select all yes.
`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver.
When this is enabled, the driver creates hardware-dependent devices
(one per each codec), and you have a raw access to the device via
these device files. For example, `hwC0D2` will be created for the
codec slot #2 of the first card (#0). For debug-tools such as
hda-verb and hda-analyzer, the hwdep device has to be enabled.
Thus, it'd be better to turn this on always.
`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the
hwdep option above. When enabled, you'll have some sysfs files under
the corresponding hwdep directory. See "HD-audio reconfiguration"
section below.
`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature.
See "Power-saving" section below.
Codec Proc-File
~~~~~~~~~~~~~~~
The codec proc-file is a treasure-chest for debugging HD-audio.
It shows most of useful information of each codec widget.
The proc file is located in /proc/asound/card*/codec#*, one file per
each codec slot. You can know the codec vendor, product id and
names, the type of each widget, capabilities and so on.
This file, however, doesn't show the jack sensing state, so far. This
is because the jack-sensing might be depending on the trigger state.
This file will be picked up by the debug tools, and also it can be fed
to the emulator as the primary codec information. See the debug tools
section below.
This proc file can be also used to check whether the generic parser is
used. When the generic parser is used, the vendor/product ID name
will appear as "Realtek ID 0262", instead of "Realtek ALC262".
HD-Audio Reconfiguration
~~~~~~~~~~~~~~~~~~~~~~~~
This is an experimental feature to allow you re-configure the HD-audio
codec dynamically without reloading the driver. The following sysfs
files are available under each codec-hwdep device directory (e.g.
/sys/class/sound/hwC0D0):
vendor_id::
Shows the 32bit codec vendor-id hex number. You can change the
vendor-id value by writing to this file.
subsystem_id::
Shows the 32bit codec subsystem-id hex number. You can change the
subsystem-id value by writing to this file.
revision_id::
Shows the 32bit codec revision-id hex number. You can change the
revision-id value by writing to this file.
afg::
Shows the AFG ID. This is read-only.
mfg::
Shows the MFG ID. This is read-only.
name::
Shows the codec name string. Can be changed by writing to this
file.
modelname::
Shows the currently set `model` option. Can be changed by writing
to this file.
init_verbs::
The extra verbs to execute at initialization. You can add a verb by
writing to this file. Pass tree numbers, nid, verb and parameter.
hints::
Shows hint strings for codec parsers for any use. Right now it's
not used.
reconfig::
Triggers the codec re-configuration. When any value is written to
this file, the driver re-initialize and parses the codec tree
again. All the changes done by the sysfs entries above are taken
into account.
clear::
Resets the codec, removes the mixer elements and PCM stuff of the
specified codec, and clear all init verbs and hints.
Power-Saving
~~~~~~~~~~~~
The power-saving is a kind of auto-suspend of the device. When the
device is inactive for a certain time, the device is automatically
turned off to save the power. The time to go down is specified via
`power_save` module option, and this option can be changed dynamically
via sysfs.
The power-saving won't work when the analog loopback is enabled on
some codecs. Make sure that you mute all unneeded signal routes when
you want the power-saving.
The power-saving feature might cause audible click noises at each
power-down/up depending on the device. Some of them might be
solvable, but some are hard, I'm afraid. Some distros such as
openSUSE enables the power-saving feature automatically when the power
cable is unplugged. Thus, if you hear noises, suspect first the
power-saving. See /sys/module/snd_hda_intel/parameters/power_save to
check the current value. If it's non-zero, the feature is turned on.
Development Tree
~~~~~~~~~~~~~~~~
The latest development codes for HD-audio are found on sound git tree:
- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
The master branch or for-next branches can be used as the main
development branches in general while the HD-audio specific patches
are committed in topic/hda branch.
If you are using the latest Linus tree, it'd be better to pull the
above GIT tree onto it. If you are using the older kernels, an easy
way to try the latest ALSA code is to build from the snapshot
tarball. There are daily tarballs and the latest snapshot tarball.
All can be built just like normal alsa-driver release packages, that
is, installed via the usual spells: configure, make and make
install(-modules). See INSTALL in the package. The snapshot tarballs
are found at:
- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/
Sending a Bug Report
~~~~~~~~~~~~~~~~~~~~
If any model or module options don't work for your device, it's time
to send a bug report to the developers. Give the following in your
bug report:
- Hardware vendor, product and model names
- Kernel version (and ALSA-driver version if you built externally)
- `alsa-info.sh` output; run with `--no-upload` option. See the
section below about alsa-info
If it's a regression, at best, send alsa-info outputs of both working
and non-working kernels. This is really helpful because we can
compare the codec registers directly.
Send a bug report either the followings:
kernel-bugzilla::
http://bugme.linux-foundation.org/
alsa-devel ML::
alsa-devel@alsa-project.org
DEBUG TOOLS
-----------
This section describes some tools available for debugging HD-audio
problems.
alsa-info
~~~~~~~~~
The script `alsa-info.sh` is a very useful tool to gather the audio
device information. You can fetch the latest version from:
- http://www.alsa-project.org/alsa-info.sh
Run this script as root, and it will gather the important information
such as the module lists, module parameters, proc file contents
including the codec proc files, mixer outputs and the control
elements. As default, it will store the information onto a web server
on alsa-project.org. But, if you send a bug report, it'd be better to
run with `--no-upload` option, and attach the generated file.
There are some other useful options. See `--help` option output for
details.
hda-verb
~~~~~~~~
hda-verb is a tiny program that allows you to access the HD-audio
codec directly. You can execute a raw HD-audio codec verb with this.
This program accesses the hwdep device, thus you need to enable the
kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand.
The hda-verb program takes four arguments: the hwdep device file, the
widget NID, the verb and the parameter. When you access to the codec
on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first
argument, typically. (However, the real path name depends on the
system.)
The second parameter is the widget number-id to access. The third
parameter can be either a hex/digit number or a string corresponding
to a verb. Similarly, the last parameter is the value to write, or
can be a string for the parameter type.
------------------------------------------------------------------------
% hda-verb /dev/snd/hwC0D0 0x12 0x701 2
nid = 0x12, verb = 0x701, param = 0x2
value = 0x0
% hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID
nid = 0x0, verb = 0xf00, param = 0x0
value = 0x10ec0262
% hda-verb /dev/snd/hwC0D0 2 set_a 0xb080
nid = 0x2, verb = 0x300, param = 0xb080
value = 0x0
------------------------------------------------------------------------
Although you can issue any verbs with this program, the driver state
won't be always updated. For example, the volume values are usually
cached in the driver, and thus changing the widget amp value directly
via hda-verb won't change the mixer value.
The hda-verb program is found in the ftp directory:
- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
Also a git repository is available:
- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git
See README file in the tarball for more details about hda-verb
program.
hda-analyzer
~~~~~~~~~~~~
hda-analyzer provides a graphical interface to access the raw HD-audio
control, based on pyGTK2 binding. It's a more powerful version of
hda-verb. The program gives you an easy-to-use GUI stuff for showing
the widget information and adjusting the amp values, as well as the
proc-compatible output.
The hda-analyzer is a part of alsa.git repository in
alsa-project.org:
- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer
Codecgraph
~~~~~~~~~~
Codecgraph is a utility program to generate a graph and visualizes the
codec-node connection of a codec chip. It's especially useful when
you analyze or debug a codec without a proper datasheet. The program
parses the given codec proc file and converts to SVG via graphiz
program.
The tarball and GIT trees are found in the web page at:
- http://helllabs.org/codecgraph/
hda-emu
~~~~~~~
hda-emu is an HD-audio emulator. The main purpose of this program is
to debug an HD-audio codec without the real hardware. Thus, it
doesn't emulate the behavior with the real audio I/O, but it just
dumps the codec register changes and the ALSA-driver internal changes
at probing and operating the HD-audio driver.
The program requires a codec proc-file to simulate. Get a proc file
for the target codec beforehand, or pick up an example codec from the
codec proc collections in the tarball. Then, run the program with the
proc file, and the hda-emu program will start parsing the codec file
and simulates the HD-audio driver:
------------------------------------------------------------------------
% hda-emu codecs/stac9200-dell-d820-laptop
# Parsing..
hda_codec: Unknown model for STAC9200, using BIOS defaults
hda_codec: pin nid 08 bios pin config 40c003fa
....
------------------------------------------------------------------------
The program gives you only a very dumb command-line interface. You
can get a proc-file dump at the current state, get a list of control
(mixer) elements, set/get the control element value, simulate the PCM
operation, the jack plugging simulation, etc.
The package is found in:
- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
A git repository is available:
- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
See README file in the tarball for more details about hda-emu
program.

10
Documentation/sound/alsa/Procfile.txt
View File

@ -153,6 +153,16 @@ card*/codec#*
Shows the general codec information and the attribute of each
widget node.
card*/eld#*
Available for HDMI or DisplayPort interfaces.
Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink,
and describes its audio capabilities and configurations.
Some ELD fields may be modified by doing `echo name hex_value > eld#*`.
Only do this if you are sure the HDMI sink provided value is wrong.
And if that makes your HDMI audio work, please report to us so that we
can fix it in future kernel releases.
Sequencer Information
---------------------

8
Documentation/sound/alsa/soc/machine.txt
View File

@ -9,7 +9,7 @@ the audio subsystem with the kernel as a platform device and is represented by
the following struct:-
/* SoC machine */
struct snd_soc_machine {
struct snd_soc_card {
char *name;
int (*probe)(struct platform_device *pdev);
@ -67,10 +67,10 @@ static struct snd_soc_dai_link corgi_dai = {
.ops = &corgi_ops,
};
struct snd_soc_machine then sets up the machine with it's DAIs. e.g.
struct snd_soc_card then sets up the machine with it's DAIs. e.g.
/* corgi audio machine driver */
static struct snd_soc_machine snd_soc_machine_corgi = {
static struct snd_soc_card snd_soc_corgi = {
.name = "Corgi",
.dai_link = &corgi_dai,
.num_links = 1,
@ -90,7 +90,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
/* corgi audio subsystem */
static struct snd_soc_device corgi_snd_devdata = {
.machine = &snd_soc_machine_corgi,
.machine = &snd_soc_corgi,
.platform = &pxa2xx_soc_platform,
.codec_dev = &soc_codec_dev_wm8731,
.codec_data = &corgi_wm8731_setup,

2
MAINTAINERS
View File

@ -3977,7 +3977,7 @@ M: tiwai@suse.de
L: alsa-devel@alsa-project.org (subscribers-only)
S: Maintained
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
P: Liam Girdwood
M: lrg@slimlogic.co.uk
P: Mark Brown

13
arch/arm/mach-pxa/include/mach/palmasoc.h Normal file
View File

@ -0,0 +1,13 @@
#ifndef _INCLUDE_PALMASOC_H_
#define _INCLUDE_PALMASOC_H_
struct palm27x_asoc_info {
int jack_gpio;
};
#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);
#else
static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) {}
#endif
#endif

2
include/linux/input.h
View File

@ -659,6 +659,8 @@ struct input_absinfo {
#define SW_RADIO SW_RFKILL_ALL /* deprecated */
#define SW_MICROPHONE_INSERT 0x04 /* set = inserted */
#define SW_DOCK 0x05 /* set = plugged into dock */
#define SW_LINEOUT_INSERT 0x06 /* set = inserted */
#define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */
#define SW_MAX 0x0f
#define SW_CNT (SW_MAX+1)

38
include/linux/mfd/wm8350/audio.h
View File

@ -1,7 +1,7 @@
/*
* audio.h -- Audio Driver for Wolfson WM8350 PMIC
*
* Copyright 2007 Wolfson Microelectronics PLC
* Copyright 2007, 2008 Wolfson Microelectronics PLC
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
@ -70,9 +70,9 @@
#define WM8350_CODEC_ISEL_0_5 3 /* x0.5 */
#define WM8350_VMID_OFF 0
#define WM8350_VMID_500K 1
#define WM8350_VMID_100K 2
#define WM8350_VMID_10K 3
#define WM8350_VMID_300K 1
#define WM8350_VMID_50K 2
#define WM8350_VMID_5K 3
/*
* R40 (0x28) - Clock Control 1
@ -591,8 +591,38 @@
#define WM8350_IRQ_CODEC_MICSCD 41
#define WM8350_IRQ_CODEC_MICD 42
/*
* WM8350 Platform data.
*
* This must be initialised per platform for best audio performance.
* Please see WM8350 datasheet for information.
*/
struct wm8350_audio_platform_data {
int vmid_discharge_msecs; /* VMID --> OFF discharge time */
int drain_msecs; /* OFF drain time */
int cap_discharge_msecs; /* Cap ON (from OFF) discharge time */
int vmid_charge_msecs; /* vmid power up time */
u32 vmid_s_curve:2; /* vmid enable s curve speed */
u32 dis_out4:2; /* out4 discharge speed */
u32 dis_out3:2; /* out3 discharge speed */
u32 dis_out2:2; /* out2 discharge speed */
u32 dis_out1:2; /* out1 discharge speed */
u32 vroi_out4:1; /* out4 tie off */
u32 vroi_out3:1; /* out3 tie off */
u32 vroi_out2:1; /* out2 tie off */
u32 vroi_out1:1; /* out1 tie off */
u32 vroi_enable:1; /* enable tie off */
u32 codec_current_on:2; /* current level ON */
u32 codec_current_standby:2; /* current level STANDBY */
u32 codec_current_charge:2; /* codec current @ vmid charge */
};
struct snd_soc_codec;
struct wm8350_codec {
struct platform_device *pdev;
struct snd_soc_codec *codec;
struct wm8350_audio_platform_data *platform_data;
};
#endif

2
include/sound/ac97_codec.h
View File

@ -281,10 +281,12 @@
/* specific - Analog Devices */
#define AC97_AD_TEST 0x5a /* test register */
#define AC97_AD_TEST2 0x5c /* undocumented test register 2 */
#define AC97_AD_HPFD_SHIFT 12 /* High Pass Filter Disable */
#define AC97_AD_CODEC_CFG 0x70 /* codec configuration */
#define AC97_AD_JACK_SPDIF 0x72 /* Jack Sense & S/PDIF */
#define AC97_AD_SERIAL_CFG 0x74 /* Serial Configuration */
#define AC97_AD_MISC 0x76 /* Misc Control Bits */
#define AC97_AD_VREFD_SHIFT 2 /* V_REFOUT Disable (AD1888) */
/* specific - Cirrus Logic */
#define AC97_CSR_ACMODE 0x5e /* AC Mode Register */

1
include/sound/asound.h
View File

@ -575,6 +575,7 @@ enum {
#define SNDRV_TIMER_GLOBAL_SYSTEM 0
#define SNDRV_TIMER_GLOBAL_RTC 1
#define SNDRV_TIMER_GLOBAL_HPET 2
#define SNDRV_TIMER_GLOBAL_HRTIMER 3
/* info flags */
#define SNDRV_TIMER_FLG_SLAVE (1<<0) /* cannot be controlled */

28
include/sound/core.h
View File

@ -353,7 +353,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
* snd_printk - printk wrapper
* @fmt: format string
*
* Works like print() but prints the file and the line of the caller
* Works like printk() but prints the file and the line of the caller
* when configured with CONFIG_SND_VERBOSE_PRINTK.
*/
#define snd_printk(fmt, args...) \
@ -380,18 +380,40 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
printk(fmt ,##args)
#endif
/**
* snd_BUG - give a BUG warning message and stack trace
*
* Calls WARN() if CONFIG_SND_DEBUG is set.
* Ignored when CONFIG_SND_DEBUG is not set.
*/
#define snd_BUG() WARN(1, "BUG?\n")
/**
* snd_BUG_ON - debugging check macro
* @cond: condition to evaluate
*
* When CONFIG_SND_DEBUG is set, this macro evaluates the given condition,
* and call WARN() and returns the value if it's non-zero.
*
* When CONFIG_SND_DEBUG is not set, this just returns zero, and the given
* condition is ignored.
*
* NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n.
* Thus, don't put any statement that influences on the code behavior,
* such as pre/post increment, to the argument of this macro.
* If you want to evaluate and give a warning, use standard WARN_ON().
*/
#define snd_BUG_ON(cond) WARN((cond), "BUG? (%s)\n", __stringify(cond))
#else /* !CONFIG_SND_DEBUG */
#define snd_printd(fmt, args...) do { } while (0)
#define snd_BUG() do { } while (0)
static inline int __snd_bug_on(void)
static inline int __snd_bug_on(int cond)
{
return 0;
}
#define snd_BUG_ON(cond) __snd_bug_on() /* always false */
#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */
#endif /* CONFIG_SND_DEBUG */

102
include/sound/info.h
View File

@ -40,30 +40,34 @@ struct snd_info_buffer {
struct snd_info_entry;
struct snd_info_entry_text {
void (*read) (struct snd_info_entry *entry, struct snd_info_buffer *buffer);
void (*write) (struct snd_info_entry *entry, struct snd_info_buffer *buffer);
void (*read)(struct snd_info_entry *entry,
struct snd_info_buffer *buffer);
void (*write)(struct snd_info_entry *entry,
struct snd_info_buffer *buffer);
};
struct snd_info_entry_ops {
int (*open) (struct snd_info_entry *entry,
unsigned short mode, void **file_private_data);
int (*release) (struct snd_info_entry * entry,
unsigned short mode, void *file_private_data);
long (*read) (struct snd_info_entry *entry, void *file_private_data,
struct file * file, char __user *buf,
int (*open)(struct snd_info_entry *entry,
unsigned short mode, void **file_private_data);
int (*release)(struct snd_info_entry *entry,
unsigned short mode, void *file_private_data);
long (*read)(struct snd_info_entry *entry, void *file_private_data,
struct file *file, char __user *buf,
unsigned long count, unsigned long pos);
long (*write)(struct snd_info_entry *entry, void *file_private_data,
struct file *file, const char __user *buf,
unsigned long count, unsigned long pos);
long (*write) (struct snd_info_entry *entry, void *file_private_data,
struct file * file, const char __user *buf,
unsigned long count, unsigned long pos);
long long (*llseek) (struct snd_info_entry *entry, void *file_private_data,
struct file * file, long long offset, int orig);
unsigned int (*poll) (struct snd_info_entry *entry, void *file_private_data,
struct file * file, poll_table * wait);
int (*ioctl) (struct snd_info_entry *entry, void *file_private_data,
struct file * file, unsigned int cmd, unsigned long arg);
int (*mmap) (struct snd_info_entry *entry, void *file_private_data,
struct inode * inode, struct file * file,
struct vm_area_struct * vma);
long long (*llseek)(struct snd_info_entry *entry,
void *file_private_data, struct file *file,
long long offset, int orig);
unsigned int(*poll)(struct snd_info_entry *entry,
void *file_private_data, struct file *file,
poll_table *wait);
int (*ioctl)(struct snd_info_entry *entry, void *file_private_data,
struct file *file, unsigned int cmd, unsigned long arg);
int (*mmap)(struct snd_info_entry *entry, void *file_private_data,
struct inode *inode, struct file *file,
struct vm_area_struct *vma);
};
struct snd_info_entry {
@ -106,34 +110,37 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer);
static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
#endif
int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) __attribute__ ((format (printf, 2, 3)));
int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \
__attribute__ ((format (printf, 2, 3)));
int snd_info_init(void);
int snd_info_done(void);
int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len);
int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len);
char *snd_info_get_str(char *dest, char *src, int len);
struct snd_info_entry *snd_info_create_module_entry(struct module * module,
struct snd_info_entry *snd_info_create_module_entry(struct module *module,
const char *name,
struct snd_info_entry * parent);
struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card,
struct snd_info_entry *parent);
struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
const char *name,
struct snd_info_entry * parent);
void snd_info_free_entry(struct snd_info_entry * entry);
int snd_info_store_text(struct snd_info_entry * entry);
int snd_info_restore_text(struct snd_info_entry * entry);
struct snd_info_entry *parent);
void snd_info_free_entry(struct snd_info_entry *entry);
int snd_info_store_text(struct snd_info_entry *entry);
int snd_info_restore_text(struct snd_info_entry *entry);
int snd_info_card_create(struct snd_card * card);
int snd_info_card_register(struct snd_card * card);
int snd_info_card_free(struct snd_card * card);
void snd_info_card_disconnect(struct snd_card * card);
int snd_info_register(struct snd_info_entry * entry);
int snd_info_card_create(struct snd_card *card);
int snd_info_card_register(struct snd_card *card);
int snd_info_card_free(struct snd_card *card);
void snd_info_card_disconnect(struct snd_card *card);
void snd_info_card_id_change(struct snd_card *card);
int snd_info_register(struct snd_info_entry *entry);
/* for card drivers */
int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp);
int snd_card_proc_new(struct snd_card *card, const char *name,
struct snd_info_entry **entryp);
static inline void snd_info_set_text_ops(struct snd_info_entry *entry,
void *private_data,
void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
void *private_data,
void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
{
entry->private_data = private_data;
entry->c.text.read = read;
@ -146,21 +153,22 @@ int snd_info_check_reserved_words(const char *str);
#define snd_seq_root NULL
#define snd_oss_root NULL
static inline int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) { return 0; }
static inline int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) { return 0; }
static inline int snd_info_init(void) { return 0; }
static inline int snd_info_done(void) { return 0; }
static inline int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len) { return 0; }
static inline int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { return 0; }
static inline char *snd_info_get_str(char *dest, char *src, int len) { return NULL; }
static inline struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry * parent) { return NULL; }
static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card, const char *name, struct snd_info_entry * parent) { return NULL; }
static inline void snd_info_free_entry(struct snd_info_entry * entry) { ; }
static inline struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent) { return NULL; }
static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent) { return NULL; }
static inline void snd_info_free_entry(struct snd_info_entry *entry) { ; }
static inline int snd_info_card_create(struct snd_card * card) { return 0; }
static inline int snd_info_card_register(struct snd_card * card) { return 0; }
static inline int snd_info_card_free(struct snd_card * card) { return 0; }
static inline void snd_info_card_disconnect(struct snd_card * card) { }
static inline int snd_info_register(struct snd_info_entry * entry) { return 0; }
static inline int snd_info_card_create(struct snd_card *card) { return 0; }
static inline int snd_info_card_register(struct snd_card *card) { return 0; }
static inline int snd_info_card_free(struct snd_card *card) { return 0; }
static inline void snd_info_card_disconnect(struct snd_card *card) { }
static inline void snd_info_card_id_change(struct snd_card *card) { }
static inline int snd_info_register(struct snd_info_entry *entry) { return 0; }
static inline int snd_card_proc_new(struct snd_card *card, const char *name,
struct snd_info_entry **entryp) { return -EINVAL; }

2
include/sound/jack.h
View File

@ -35,6 +35,8 @@ enum snd_jack_types {
SND_JACK_HEADPHONE = 0x0001,
SND_JACK_MICROPHONE = 0x0002,
SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE,
SND_JACK_LINEOUT = 0x0004,
SND_JACK_MECHANICAL = 0x0008, /* If detected separately */
};
struct snd_jack {

18
include/sound/l3.h Normal file
View File

@ -0,0 +1,18 @@
#ifndef _L3_H_
#define _L3_H_ 1
struct l3_pins {
void (*setdat)(int);
void (*setclk)(int);
void (*setmode)(int);
int data_hold;
int data_setup;
int clock_high;
int mode_hold;
int mode;
int mode_setup;
};
int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
#endif

14
include/sound/s3c24xx_uda134x.h Normal file
View File

@ -0,0 +1,14 @@
#ifndef _S3C24XX_UDA134X_H_
#define _S3C24XX_UDA134X_H_ 1
#include <sound/uda134x.h>
struct s3c24xx_uda134x_platform_data {
int l3_clk;
int l3_mode;
int l3_data;
void (*power) (int);
int model;
};
#endif

231
include/sound/soc-dai.h Normal file
View File

@ -0,0 +1,231 @@
/*
* linux/sound/soc-dai.h -- ALSA SoC Layer
*
* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Digital Audio Interface (DAI) API.
*/
#ifndef __LINUX_SND_SOC_DAI_H
#define __LINUX_SND_SOC_DAI_H
#include <linux/list.h>
struct snd_pcm_substream;
/*
* DAI hardware audio formats.
*
* Describes the physical PCM data formating and clocking. Add new formats
* to the end.
*/
#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
/*
* DAI Clock gating.
*
* DAI bit clocks can be be gated (disabled) when not the DAI is not
* sending or receiving PCM data in a frame. This can be used to save power.
*/
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
/*
* DAI Left/Right Clocks.
*
* Specifies whether the DAI can support different samples for similtanious
* playback and capture. This usually requires a seperate physical frame
* clock for playback and capture.
*/
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
/*
* TDM
*
* Time Division Multiplexing. Allows PCM data to be multplexed with other
* data on the DAI.
*/
#define SND_SOC_DAIFMT_TDM (1 << 6)
/*
* DAI hardware signal inversions.
*
* Specifies whether the DAI can also support inverted clocks for the specified
* format.
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
/*
* DAI hardware clock masters.
*
* This is wrt the codec, the inverse is true for the interface
* i.e. if the codec is clk and frm master then the interface is
* clk and frame slave.
*/
#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */
#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
#define SND_SOC_DAIFMT_INV_MASK 0x0f00
#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
/*
* Master Clock Directions
*/
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
/* Digital Audio Interface registration */
int snd_soc_register_dai(struct snd_soc_dai *dai);
void snd_soc_unregister_dai(struct snd_soc_dai *dai);
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
/* Digital Audio Interface clocking API.*/
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir);
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
int div_id, int div);
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out);
/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int mask, int slots);
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
/*
* Digital Audio Interface.
*
* Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
* operations an capabilities. Codec and platfom drivers will register a this
* structure for every DAI they have.
*
* This structure covers the clocking, formating and ALSA operations for each
* interface a
*/
struct snd_soc_dai_ops {
/*
* DAI clocking configuration, all optional.
* Called by soc_card drivers, normally in their hw_params.
*/
int (*set_sysclk)(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir);
int (*set_pll)(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/*
* DAI format configuration
* Called by soc_card drivers, normally in their hw_params.
*/
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int mask, int slots);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
* DAI digital mute - optional.
* Called by soc-core to minimise any pops.
*/
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
/*
* ALSA PCM audio operations - all optional.
* Called by soc-core during audio PCM operations.
*/
int (*startup)(struct snd_pcm_substream *,
struct snd_soc_dai *);
void (*shutdown)(struct snd_pcm_substream *,
struct snd_soc_dai *);
int (*hw_params)(struct snd_pcm_substream *,
struct snd_pcm_hw_params *, struct snd_soc_dai *);
int (*hw_free)(struct snd_pcm_substream *,
struct snd_soc_dai *);
int (*prepare)(struct snd_pcm_substream *,
struct snd_soc_dai *);
int (*trigger)(struct snd_pcm_substream *, int,
struct snd_soc_dai *);
};
/*
* Digital Audio Interface runtime data.
*
* Holds runtime data for a DAI.
*/
struct snd_soc_dai {
/* DAI description */
char *name;
unsigned int id;
int ac97_control;
struct device *dev;
/* DAI callbacks */
int (*probe)(struct platform_device *pdev,
struct snd_soc_dai *dai);
void (*remove)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*suspend)(struct snd_soc_dai *dai);
int (*resume)(struct snd_soc_dai *dai);
/* ops */
struct snd_soc_dai_ops ops;
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
/* DAI runtime info */
struct snd_pcm_runtime *runtime;
struct snd_soc_codec *codec;
unsigned int active;
unsigned char pop_wait:1;
void *dma_data;
/* DAI private data */
void *private_data;
/* parent codec/platform */
union {
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
};
struct list_head list;
};
#endif

2
include/sound/soc-dapm.h
View File

@ -221,8 +221,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
int num);
/* dapm path setup */
int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
const char *sink_name, const char *control_name, const char *src_name);
int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
void snd_soc_dapm_free(struct snd_soc_device *socdev);
int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,

206
include/sound/soc.h
View File

@ -21,8 +21,6 @@
#include <sound/control.h>
#include <sound/ac97_codec.h>
#define SND_SOC_VERSION "0.13.2"
/*
* Convenience kcontrol builders
*/
@ -145,105 +143,31 @@ enum snd_soc_bias_level {
SND_SOC_BIAS_OFF,
};
/*
* Digital Audio Interface (DAI) types
*/
#define SND_SOC_DAI_AC97 0x1
#define SND_SOC_DAI_I2S 0x2
#define SND_SOC_DAI_PCM 0x4
#define SND_SOC_DAI_AC97_BUS 0x8 /* for custom i.e. non ac97_codec.c */
/*
* DAI hardware audio formats
*/
#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right justified mode */
#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM or LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM or LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
/*
* DAI Gating
*/
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */
/*
* DAI Sync
* Synchronous LR (Left Right) clocks and Frame signals.
*/
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
/*
* TDM
*/
#define SND_SOC_DAIFMT_TDM (1 << 6)
/*
* DAI hardware signal inversions
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */
#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
/*
* DAI hardware clock masters
* This is wrt the codec, the inverse is true for the interface
* i.e. if the codec is clk and frm master then the interface is
* clk and frame slave.
*/
#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */
#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
#define SND_SOC_DAIFMT_INV_MASK 0x0f00
#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
/*
* Master Clock Directions
*/
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
/*
* AC97 codec ID's bitmask
*/
#define SND_SOC_DAI_AC97_ID0 (1 << 0)
#define SND_SOC_DAI_AC97_ID1 (1 << 1)
#define SND_SOC_DAI_AC97_ID2 (1 << 2)
#define SND_SOC_DAI_AC97_ID3 (1 << 3)
struct snd_soc_device;
struct snd_soc_pcm_stream;
struct snd_soc_ops;
struct snd_soc_dai_mode;
struct snd_soc_pcm_runtime;
struct snd_soc_dai;
struct snd_soc_platform;
struct snd_soc_codec;
struct snd_soc_machine_config;
struct soc_enum;
struct snd_soc_ac97_ops;
struct snd_soc_clock_info;
typedef int (*hw_write_t)(void *,const char* ,int);
typedef int (*hw_read_t)(void *,char* ,int);
extern struct snd_ac97_bus_ops soc_ac97_ops;
int snd_soc_register_platform(struct snd_soc_platform *platform);
void snd_soc_unregister_platform(struct snd_soc_platform *platform);
int snd_soc_register_codec(struct snd_soc_codec *codec);
void snd_soc_unregister_codec(struct snd_soc_codec *codec);
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
int snd_soc_register_card(struct snd_soc_device *socdev);
int snd_soc_init_card(struct snd_soc_device *socdev);
/* set runtime hw params */
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
@ -263,27 +187,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
struct snd_ac97_bus_ops *ops, int num);
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
/* Digital Audio Interface clocking API.*/
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir);
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
int div_id, int div);
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out);
/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int mask, int slots);
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
/*
*Controls
*/
@ -341,66 +244,14 @@ struct snd_soc_ops {
int (*trigger)(struct snd_pcm_substream *, int);
};
/* ASoC DAI ops */
struct snd_soc_dai_ops {
/* DAI clocking configuration */
int (*set_sysclk)(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir);
int (*set_pll)(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/* DAI format configuration */
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int mask, int slots);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/* digital mute */
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
};
/* SoC DAI (Digital Audio Interface) */
struct snd_soc_dai {
/* DAI description */
char *name;
unsigned int id;
unsigned char type;
/* DAI callbacks */
int (*probe)(struct platform_device *pdev,
struct snd_soc_dai *dai);
void (*remove)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*suspend)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*resume)(struct platform_device *pdev,
struct snd_soc_dai *dai);
/* ops */
struct snd_soc_ops ops;
struct snd_soc_dai_ops dai_ops;
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
/* DAI runtime info */
struct snd_pcm_runtime *runtime;
struct snd_soc_codec *codec;
unsigned int active;
unsigned char pop_wait:1;
void *dma_data;
/* DAI private data */
void *private_data;
};
/* SoC Audio Codec */
struct snd_soc_codec {
char *name;
struct module *owner;
struct mutex mutex;
struct device *dev;
struct list_head list;
/* callbacks */
int (*set_bias_level)(struct snd_soc_codec *,
@ -426,6 +277,7 @@ struct snd_soc_codec {
short reg_cache_step;
/* dapm */
u32 pop_time;
struct list_head dapm_widgets;
struct list_head dapm_paths;
enum snd_soc_bias_level bias_level;
@ -435,6 +287,11 @@ struct snd_soc_codec {
/* codec DAI's */
struct snd_soc_dai *dai;
unsigned int num_dai;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
#endif
};
/* codec device */
@ -448,13 +305,12 @@ struct snd_soc_codec_device {
/* SoC platform interface */
struct snd_soc_platform {
char *name;
struct list_head list;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
int (*suspend)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*resume)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*suspend)(struct snd_soc_dai *dai);
int (*resume)(struct snd_soc_dai *dai);
/* pcm creation and destruction */
int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
@ -484,9 +340,14 @@ struct snd_soc_dai_link {
struct snd_pcm *pcm;
};
/* SoC machine */
struct snd_soc_machine {
/* SoC card */
struct snd_soc_card {
char *name;
struct device *dev;
struct list_head list;
int instantiated;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
@ -499,23 +360,26 @@ struct snd_soc_machine {
int (*resume_post)(struct platform_device *pdev);
/* callbacks */
int (*set_bias_level)(struct snd_soc_machine *,
int (*set_bias_level)(struct snd_soc_card *,
enum snd_soc_bias_level level);
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;
struct snd_soc_device *socdev;
struct snd_soc_platform *platform;
struct delayed_work delayed_work;
struct work_struct deferred_resume_work;
};
/* SoC Device - the audio subsystem */
struct snd_soc_device {
struct device *dev;
struct snd_soc_machine *machine;
struct snd_soc_platform *platform;
struct snd_soc_card *card;
struct snd_soc_codec *codec;
struct snd_soc_codec_device *codec_dev;
struct delayed_work delayed_work;
struct work_struct deferred_resume_work;
void *codec_data;
};
@ -542,4 +406,6 @@ struct soc_enum {
void *dapm;
};
#include <sound/soc-dai.h>
#endif

26
include/sound/uda134x.h Normal file
View File

@ -0,0 +1,26 @@
/*
* uda134x.h -- UDA134x ALSA SoC Codec driver
*
* Copyright 2007 Dension Audio Systems Ltd.
* Author: Zoltan Devai
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef _UDA134X_H
#define _UDA134X_H
#include <sound/l3.h>
struct uda134x_platform_data {
struct l3_pins l3;
void (*power) (int);
int model;
#define UDA134X_UDA1340 1
#define UDA134X_UDA1341 2
#define UDA134X_UDA1344 3
};
#endif /* _UDA134X_H */

2
include/sound/version.h
View File

@ -1,3 +1,3 @@
/* include/version.h */
#define CONFIG_SND_VERSION "1.0.18rc3"
#define CONFIG_SND_VERSION "1.0.18a"
#define CONFIG_SND_DATE ""

1
sound/ac97_bus.c
View File

@ -15,6 +15,7 @@
#include <linux/init.h>
#include <linux/device.h>
#include <linux/string.h>
#include <sound/ac97_codec.h>
/*
* Let drivers decide whether they want to support given codec from their

4
sound/aoa/codecs/Makefile
View File

@ -1,3 +1,7 @@
snd-aoa-codec-onyx-objs := onyx.o
snd-aoa-codec-tas-objs := tas.o
snd-aoa-codec-toonie-objs := toonie.o
obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o

12
sound/aoa/codecs/snd-aoa-codec-onyx.c → sound/aoa/codecs/onyx.c
View File

@ -37,7 +37,7 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa");
#include "snd-aoa-codec-onyx.h"
#include "onyx.h"
#include "../aoa.h"
#include "../soundbus/soundbus.h"
@ -292,7 +292,7 @@ static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol,
static struct snd_kcontrol_new capture_source_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* If we name this 'Input Source', it properly shows up in
* alsamixer as a selection, * but it's shown under the
* alsamixer as a selection, * but it's shown under the
* 'Playback' category.
* If I name it 'Capture Source', it shows up in strange
* ways (two bools of which one can be selected at a
@ -477,7 +477,7 @@ static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol,
ucontrol->value.iec958.status[3] = 0x3f;
ucontrol->value.iec958.status[4] = 0x0f;
return 0;
}
@ -682,7 +682,7 @@ static int onyx_usable(struct codec_info_item *cii,
onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
spdif_enabled = !!(v & ONYX_SPDIF_ENABLE);
onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
analog_enabled =
analog_enabled =
(v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT))
!= (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT);
mutex_unlock(&onyx->mutex);
@ -882,7 +882,7 @@ static int onyx_init_codec(struct aoa_codec *codec)
msleep(1);
onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
msleep(1);
if (onyx_register_init(onyx)) {
printk(KERN_ERR PFX "failed to initialise onyx registers\n");
return -ENODEV;
@ -1069,7 +1069,7 @@ static int onyx_i2c_attach(struct i2c_adapter *adapter)
/* if that didn't work, try desperate mode for older
* machines that have stuff missing from the device tree */
if (!of_device_is_compatible(busnode, "k2-i2c"))
return -ENODEV;

0
sound/aoa/codecs/snd-aoa-codec-onyx.h → sound/aoa/codecs/onyx.h
View File

0
sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h → sound/aoa/codecs/tas-basstreble.h
View File

0
sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h → sound/aoa/codecs/tas-gain-table.h
View File

8
sound/aoa/codecs/snd-aoa-codec-tas.c → sound/aoa/codecs/tas.c
View File

@ -71,9 +71,9 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("tas codec driver for snd-aoa");
#include "snd-aoa-codec-tas.h"
#include "snd-aoa-codec-tas-gain-table.h"
#include "snd-aoa-codec-tas-basstreble.h"
#include "tas.h"
#include "tas-gain-table.h"
#include "tas-basstreble.h"
#include "../aoa.h"
#include "../soundbus/soundbus.h"
@ -880,7 +880,7 @@ static void tas_exit_codec(struct aoa_codec *codec)
return;
tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
}
static struct i2c_driver tas_driver;

0
sound/aoa/codecs/snd-aoa-codec-tas.h → sound/aoa/codecs/tas.h
View File

2
sound/aoa/codecs/snd-aoa-codec-toonie.c → sound/aoa/codecs/toonie.c
View File

@ -131,7 +131,7 @@ static int __init toonie_init(void)
toonie->codec.owner = THIS_MODULE;
toonie->codec.init = toonie_init_codec;
toonie->codec.exit = toonie_exit_codec;
if (aoa_codec_register(&toonie->codec)) {
kfree(toonie);
return -EINVAL;

8
sound/aoa/core/Makefile
View File

@ -1,5 +1,5 @@
obj-$(CONFIG_SND_AOA) += snd-aoa.o
snd-aoa-objs := snd-aoa-core.o \
snd-aoa-alsa.o \
snd-aoa-gpio-pmf.o \
snd-aoa-gpio-feature.o
snd-aoa-objs := core.o \
alsa.o \
gpio-pmf.o \
gpio-feature.o

4
sound/aoa/core/snd-aoa-alsa.c → sound/aoa/core/alsa.c
View File

@ -6,7 +6,7 @@
* GPL v2, can be found in COPYING.
*/
#include <linux/module.h>
#include "snd-aoa-alsa.h"
#include "alsa.h"
static int index = -1;
module_param(index, int, 0444);
@ -64,7 +64,7 @@ int aoa_snd_device_new(snd_device_type_t type,
{
struct snd_card *card = aoa_get_card();
int err;
if (!card) return -ENOMEM;
err = snd_device_new(card, type, device_data, ops);

0
sound/aoa/core/snd-aoa-alsa.h → sound/aoa/core/alsa.h
View File

2
sound/aoa/core/snd-aoa-core.c → sound/aoa/core/core.c
View File

@ -10,7 +10,7 @@
#include <linux/module.h>
#include <linux/list.h>
#include "../aoa.h"
#include "snd-aoa-alsa.h"
#include "alsa.h"
MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver");
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");

2
sound/aoa/core/snd-aoa-gpio-feature.c → sound/aoa/core/gpio-feature.c
View File

@ -5,7 +5,7 @@
*
* GPL v2, can be found in COPYING.
*
* This file contains the GPIO control routines for
* This file contains the GPIO control routines for
* direct (through feature calls) access to the GPIO
* registers.
*/

0
sound/aoa/core/snd-aoa-gpio-pmf.c → sound/aoa/core/gpio-pmf.c
View File

2
sound/aoa/fabrics/Makefile
View File

@ -1 +1,3 @@
snd-aoa-fabric-layout-objs += layout.o
obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o

2
sound/aoa/fabrics/snd-aoa-fabric-layout.c → sound/aoa/fabrics/layout.c
View File

@ -66,7 +66,7 @@ struct layout {
unsigned int layout_id;
struct codec_connect_info codecs[MAX_CODECS_PER_BUS];
int flags;
/* if busname is not assigned, we use 'Master' below,
* so that our layout table doesn't need to be filled
* too much.

2
sound/aoa/soundbus/i2sbus/Makefile
View File

@ -1,2 +1,2 @@
obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o
snd-aoa-i2sbus-objs := core.o pcm.o control.o

0
sound/aoa/soundbus/i2sbus/i2sbus-control.c → sound/aoa/soundbus/i2sbus/control.c
View File

4
sound/aoa/soundbus/i2sbus/i2sbus-core.c → sound/aoa/soundbus/i2sbus/core.c
View File

@ -64,7 +64,7 @@ static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
struct dbdma_command_mem *r)
{
if (!r->space) return;
dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev,
r->size, r->space, r->bus_addr);
}
@ -247,7 +247,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
* but request_resource doesn't know about parents and
* contained resources...
*/
dev->allocated_resource[i] =
dev->allocated_resource[i] =
request_mem_region(dev->resources[i].start,
dev->resources[i].end -
dev->resources[i].start + 1,

2
sound/aoa/soundbus/i2sbus/i2sbus.h
View File

@ -18,7 +18,7 @@
#include <asm/pmac_feature.h>
#include <asm/dbdma.h>
#include "i2sbus-interface.h"
#include "interface.h"
#include "../soundbus.h"
struct i2sbus_control {

0
sound/aoa/soundbus/i2sbus/i2sbus-interface.h → sound/aoa/soundbus/i2sbus/interface.h
View File

0
sound/aoa/soundbus/i2sbus/i2sbus-pcm.c → sound/aoa/soundbus/i2sbus/pcm.c
View File

21
sound/core/Kconfig
View File

@ -95,6 +95,26 @@ config SND_SEQUENCER_OSS
this will be compiled as a module. The module will be called
snd-seq-oss.
config SND_HRTIMER
tristate "HR-timer backend support"
depends on HIGH_RES_TIMERS
select SND_TIMER
help
Say Y here to enable HR-timer backend for ALSA timer. ALSA uses
the hrtimer as a precise timing source. The ALSA sequencer code
also can use this timing source.
To compile this driver as a module, choose M here: the module
will be called snd-hrtimer.
config SND_SEQ_HRTIMER_DEFAULT
bool "Use HR-timer as default sequencer timer"
depends on SND_HRTIMER && SND_SEQUENCER
default y
help
Say Y here to use the HR-timer backend as the default sequencer
timer.
config SND_RTCTIMER
tristate "RTC Timer support"
depends on RTC
@ -114,6 +134,7 @@ config SND_RTCTIMER
config SND_SEQ_RTCTIMER_DEFAULT
bool "Use RTC as default sequencer timer"
depends on SND_RTCTIMER && SND_SEQUENCER
depends on !SND_SEQ_HRTIMER_DEFAULT
default y
help
Say Y here to use the RTC timer as the default sequencer

2
sound/core/Makefile
View File

@ -17,12 +17,14 @@ snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o
snd-rawmidi-objs := rawmidi.o
snd-timer-objs := timer.o
snd-hrtimer-objs := hrtimer.o
snd-rtctimer-objs := rtctimer.o
snd-hwdep-objs := hwdep.o
obj-$(CONFIG_SND) += snd.o
obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o
obj-$(CONFIG_SND_TIMER) += snd-timer.o
obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o
obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o
obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o
obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o

4
sound/core/device.c
View File

@ -98,7 +98,7 @@ int snd_device_free(struct snd_card *card, void *device_data)
kfree(dev);
return 0;
}
snd_printd("device free %p (from %p), not found\n", device_data,
snd_printd("device free %p (from %pF), not found\n", device_data,
__builtin_return_address(0));
return -ENXIO;
}
@ -135,7 +135,7 @@ int snd_device_disconnect(struct snd_card *card, void *device_data)
}
return 0;
}
snd_printd("device disconnect %p (from %p), not found\n", device_data,
snd_printd("device disconnect %p (from %pF), not found\n", device_data,
__builtin_return_address(0));
return -ENXIO;
}

155
sound/core/hrtimer.c Normal file
View File

@ -0,0 +1,155 @@
/*
* ALSA timer back-end using hrtimer
* Copyright (C) 2008 Takashi Iwai
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/hrtimer.h>
#include <sound/core.h>
#include <sound/timer.h>
MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
MODULE_DESCRIPTION("ALSA hrtimer backend");
MODULE_LICENSE("GPL");
MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_HRTIMER));
#define NANO_SEC 1000000000UL /* 10^9 in sec */
static unsigned int resolution;
struct snd_hrtimer {
struct snd_timer *timer;
struct hrtimer hrt;
};
static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
{
struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
struct snd_timer *t = stime->timer;
hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
snd_timer_interrupt(stime->timer, t->sticks);
return HRTIMER_RESTART;
}
static int snd_hrtimer_open(struct snd_timer *t)
{
struct snd_hrtimer *stime;
stime = kmalloc(sizeof(*stime), GFP_KERNEL);
if (!stime)
return -ENOMEM;
hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
stime->timer = t;
stime->hrt.cb_mode = HRTIMER_CB_IRQSAFE_UNLOCKED;
stime->hrt.function = snd_hrtimer_callback;
t->private_data = stime;
return 0;
}
static int snd_hrtimer_close(struct snd_timer *t)
{
struct snd_hrtimer *stime = t->private_data;
if (stime) {
hrtimer_cancel(&stime->hrt);
kfree(stime);
t->private_data = NULL;
}
return 0;
}
static int snd_hrtimer_start(struct snd_timer *t)
{
struct snd_hrtimer *stime = t->private_data;
hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution),
HRTIMER_MODE_REL);
return 0;
}
static int snd_hrtimer_stop(struct snd_timer *t)
{
struct snd_hrtimer *stime = t->private_data;
hrtimer_cancel(&stime->hrt);
return 0;
}
static struct snd_timer_hardware hrtimer_hw = {
.flags = SNDRV_TIMER_HW_AUTO,
.open = snd_hrtimer_open,
.close = snd_hrtimer_close,
.start = snd_hrtimer_start,
.stop = snd_hrtimer_stop,
};
/*
* entry functions
*/
static struct snd_timer *mytimer;
static int __init snd_hrtimer_init(void)
{
struct snd_timer *timer;
struct timespec tp;
int err;
hrtimer_get_res(CLOCK_MONOTONIC, &tp);
if (tp.tv_sec > 0 || !tp.tv_nsec) {
snd_printk(KERN_ERR
"snd-hrtimer: Invalid resolution %u.%09u",
(unsigned)tp.tv_sec, (unsigned)tp.tv_nsec);
return -EINVAL;
}
resolution = tp.tv_nsec;
/* Create a new timer and set up the fields */
err = snd_timer_global_new("hrtimer", SNDRV_TIMER_GLOBAL_HRTIMER,
&timer);
if (err < 0)
return err;
timer->module = THIS_MODULE;
strcpy(timer->name, "HR timer");
timer->hw = hrtimer_hw;
timer->hw.resolution = resolution;
timer->hw.ticks = NANO_SEC / resolution;
err = snd_timer_global_register(timer);
if (err < 0) {
snd_timer_global_free(timer);
return err;
}
mytimer = timer; /* remember this */
return 0;
}
static void __exit snd_hrtimer_exit(void)
{
if (mytimer) {
snd_timer_global_free(mytimer);
mytimer = NULL;
}
}
module_init(snd_hrtimer_init);
module_exit(snd_hrtimer_exit);

17
sound/core/info.c
View File

@ -652,6 +652,23 @@ int snd_info_card_register(struct snd_card *card)
return 0;
}
/*
* called on card->id change
*/
void snd_info_card_id_change(struct snd_card *card)
{
mutex_lock(&info_mutex);
if (card->proc_root_link) {
snd_remove_proc_entry(snd_proc_root, card->proc_root_link);
card->proc_root_link = NULL;
}
if (strcmp(card->id, card->proc_root->name))
card->proc_root_link = proc_symlink(card->id,
snd_proc_root,
card->proc_root->name);
mutex_unlock(&info_mutex);
}
/*
* de-register the card proc file
* called from init.c

71
sound/core/init.c
View File

@ -533,6 +533,65 @@ static void choose_default_id(struct snd_card *card)
}
}
#ifndef CONFIG_SYSFS_DEPRECATED
static ssize_t
card_id_show_attr(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_card *card = dev_get_drvdata(dev);
return snprintf(buf, PAGE_SIZE, "%s\n", card ? card->id : "(null)");
}
static ssize_t
card_id_store_attr(struct device *dev, struct device_attribute *attr,
const char *buf, size_t count)
{
struct snd_card *card = dev_get_drvdata(dev);
char buf1[sizeof(card->id)];
size_t copy = count > sizeof(card->id) - 1 ?
sizeof(card->id) - 1 : count;
size_t idx;
int c;
for (idx = 0; idx < copy; idx++) {
c = buf[idx];
if (!isalnum(c) && c != '_' && c != '-')
return -EINVAL;
}
memcpy(buf1, buf, copy);
buf1[copy] = '\0';
mutex_lock(&snd_card_mutex);
if (!snd_info_check_reserved_words(buf1)) {
__exist:
mutex_unlock(&snd_card_mutex);
return -EEXIST;
}
for (idx = 0; idx < snd_ecards_limit; idx++) {
if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1))
goto __exist;
}
strcpy(card->id, buf1);
snd_info_card_id_change(card);
mutex_unlock(&snd_card_mutex);
return count;
}
static struct device_attribute card_id_attrs =
__ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr);
static ssize_t
card_number_show_attr(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_card *card = dev_get_drvdata(dev);
return snprintf(buf, PAGE_SIZE, "%i\n", card ? card->number : -1);
}
static struct device_attribute card_number_attrs =
__ATTR(number, S_IRUGO, card_number_show_attr, NULL);
#endif /* CONFIG_SYSFS_DEPRECATED */
/**
* snd_card_register - register the soundcard
* @card: soundcard structure
@ -553,7 +612,7 @@ int snd_card_register(struct snd_card *card)
#ifndef CONFIG_SYSFS_DEPRECATED
if (!card->card_dev) {
card->card_dev = device_create(sound_class, card->dev,
MKDEV(0, 0), NULL,
MKDEV(0, 0), card,
"card%i", card->number);
if (IS_ERR(card->card_dev))
card->card_dev = NULL;
@ -575,6 +634,16 @@ int snd_card_register(struct snd_card *card)
#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
if (snd_mixer_oss_notify_callback)
snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER);
#endif
#ifndef CONFIG_SYSFS_DEPRECATED
if (card->card_dev) {
err = device_create_file(card->card_dev, &card_id_attrs);
if (err < 0)
return err;
err = device_create_file(card->card_dev, &card_number_attrs);
if (err < 0)
return err;
}
#endif
return 0;
}

15
sound/core/jack.c
View File

@ -34,6 +34,7 @@ static int snd_jack_dev_free(struct snd_device *device)
else
input_free_device(jack->input_dev);
kfree(jack->id);
kfree(jack);
return 0;
@ -87,7 +88,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
if (jack == NULL)
return -ENOMEM;
jack->id = id;
jack->id = kstrdup(id, GFP_KERNEL);
jack->input_dev = input_allocate_device();
if (jack->input_dev == NULL) {
@ -102,9 +103,15 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
if (type & SND_JACK_HEADPHONE)
input_set_capability(jack->input_dev, EV_SW,
SW_HEADPHONE_INSERT);
if (type & SND_JACK_LINEOUT)
input_set_capability(jack->input_dev, EV_SW,
SW_LINEOUT_INSERT);
if (type & SND_JACK_MICROPHONE)
input_set_capability(jack->input_dev, EV_SW,
SW_MICROPHONE_INSERT);
if (type & SND_JACK_MECHANICAL)
input_set_capability(jack->input_dev, EV_SW,
SW_JACK_PHYSICAL_INSERT);
err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops);
if (err < 0)
@ -153,9 +160,15 @@ void snd_jack_report(struct snd_jack *jack, int status)
if (jack->type & SND_JACK_HEADPHONE)
input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT,
status & SND_JACK_HEADPHONE);
if (jack->type & SND_JACK_LINEOUT)
input_report_switch(jack->input_dev, SW_LINEOUT_INSERT,
status & SND_JACK_LINEOUT);
if (jack->type & SND_JACK_MICROPHONE)
input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT,
status & SND_JACK_MICROPHONE);
if (jack->type & SND_JACK_MECHANICAL)
input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT,
status & SND_JACK_MECHANICAL);
input_sync(jack->input_dev);
}

4
sound/core/rawmidi.c
View File

@ -151,7 +151,7 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs
if (!substream->opened)
return;
if (up) {
tasklet_hi_schedule(&substream->runtime->tasklet);
tasklet_schedule(&substream->runtime->tasklet);
} else {
tasklet_kill(&substream->runtime->tasklet);
substream->ops->trigger(substream, 0);
@ -908,7 +908,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
}
if (result > 0) {
if (runtime->event)
tasklet_hi_schedule(&runtime->tasklet);
tasklet_schedule(&runtime->tasklet);
else if (snd_rawmidi_ready(substream))
wake_up(&runtime->sleep);
}

2
sound/core/rtctimer.c
View File

@ -118,7 +118,7 @@ static void rtctimer_tasklet(unsigned long data)
*/
static void rtctimer_interrupt(void *private_data)
{
tasklet_hi_schedule(private_data);
tasklet_schedule(private_data);
}

4
sound/core/seq/seq.c
View File

@ -43,7 +43,9 @@ int seq_default_timer_class = SNDRV_TIMER_CLASS_GLOBAL;
int seq_default_timer_sclass = SNDRV_TIMER_SCLASS_NONE;
int seq_default_timer_card = -1;
int seq_default_timer_device =
#ifdef CONFIG_SND_SEQ_RTCTIMER_DEFAULT
#ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT
SNDRV_TIMER_GLOBAL_HRTIMER
#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT)
SNDRV_TIMER_GLOBAL_RTC
#else
SNDRV_TIMER_GLOBAL_SYSTEM

2
sound/core/timer.c
View File

@ -743,7 +743,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
spin_unlock_irqrestore(&timer->lock, flags);
if (use_tasklet)
tasklet_hi_schedule(&timer->task_queue);
tasklet_schedule(&timer->task_queue);
}
/*

2
sound/drivers/Kconfig
View File

@ -163,7 +163,7 @@ config SND_ML403_AC97CR
config SND_AC97_POWER_SAVE
bool "AC97 Power-Saving Mode"
depends on SND_AC97_CODEC && EXPERIMENTAL
depends on SND_AC97_CODEC
default n
help
Say Y here to enable the aggressive power-saving support of

8
sound/drivers/pcsp/pcsp.c
View File

@ -96,7 +96,7 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev)
return -EINVAL;
hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
pcsp_chip.timer.cb_mode = HRTIMER_CB_SOFTIRQ;
pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE_UNLOCKED;
pcsp_chip.timer.function = pcsp_do_timer;
card = snd_card_new(index, id, THIS_MODULE, 0);
@ -188,10 +188,8 @@ static int __devexit pcsp_remove(struct platform_device *dev)
static void pcsp_stop_beep(struct snd_pcsp *chip)
{
spin_lock_irq(&chip->substream_lock);
if (!chip->playback_substream)
pcspkr_stop_sound();
spin_unlock_irq(&chip->substream_lock);
pcsp_sync_stop(chip);
pcspkr_stop_sound();
}
#ifdef CONFIG_PM

3
sound/drivers/pcsp/pcsp.h
View File

@ -62,6 +62,8 @@ struct snd_pcsp {
unsigned short port, irq, dma;
spinlock_t substream_lock;
struct snd_pcm_substream *playback_substream;
unsigned int fmt_size;
unsigned int is_signed;
size_t playback_ptr;
size_t period_ptr;
atomic_t timer_active;
@ -77,6 +79,7 @@ struct snd_pcsp {
extern struct snd_pcsp pcsp_chip;
extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle);
extern void pcsp_sync_stop(struct snd_pcsp *chip);
extern int snd_pcsp_new_pcm(struct snd_pcsp *chip);
extern int snd_pcsp_new_mixer(struct snd_pcsp *chip);

168
sound/drivers/pcsp/pcsp_lib.c
View File

@ -8,6 +8,7 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/interrupt.h>
#include <sound/pcm.h>
#include <asm/io.h>
#include "pcsp.h"
@ -19,61 +20,57 @@ MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround "
#define DMIX_WANTS_S16 1
enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
/*
* Call snd_pcm_period_elapsed in a tasklet
* This avoids spinlock messes and long-running irq contexts
*/
static void pcsp_call_pcm_elapsed(unsigned long priv)
{
if (atomic_read(&pcsp_chip.timer_active)) {
struct snd_pcm_substream *substream;
substream = pcsp_chip.playback_substream;
if (substream)
snd_pcm_period_elapsed(substream);
}
}
static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0);
/* write the port and returns the next expire time in ns;
* called at the trigger-start and in hrtimer callback
*/
static unsigned long pcsp_timer_update(struct hrtimer *handle)
{
unsigned char timer_cnt, val;
int fmt_size, periods_elapsed;
u64 ns;
size_t period_bytes, buffer_bytes;
struct snd_pcm_substream *substream;
struct snd_pcm_runtime *runtime;
struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
unsigned long flags;
if (chip->thalf) {
outb(chip->val61, 0x61);
chip->thalf = 0;
if (!atomic_read(&chip->timer_active))
return HRTIMER_NORESTART;
hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
ktime_set(0, chip->ns_rem));
return HRTIMER_RESTART;
return 0;
return chip->ns_rem;
}
spin_lock_irq(&chip->substream_lock);
/* Takashi Iwai says regarding this extra lock:
If the irq handler handles some data on the DMA buffer, it should
do snd_pcm_stream_lock().
That protects basically against all races among PCM callbacks, yes.
However, there are two remaining issues:
1. The substream pointer you try to lock isn't protected _before_
this lock yet.
2. snd_pcm_period_elapsed() itself acquires the lock.
The requirement of another lock is because of 1. When you get
chip->playback_substream, it's not protected.
Keeping this lock while snd_pcm_period_elapsed() assures the substream
is still protected (at least, not released). And the other status is
handled properly inside snd_pcm_stream_lock() in
snd_pcm_period_elapsed().
*/
if (!chip->playback_substream)
goto exit_nr_unlock1;
substream = chip->playback_substream;
snd_pcm_stream_lock(substream);
if (!atomic_read(&chip->timer_active))
goto exit_nr_unlock2;
return 0;
substream = chip->playback_substream;
if (!substream)
return 0;
runtime = substream->runtime;
fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3;
/* assume it is mono! */
val = runtime->dma_area[chip->playback_ptr + fmt_size - 1];
if (snd_pcm_format_signed(runtime->format))
val = runtime->dma_area[chip->playback_ptr + chip->fmt_size - 1];
if (chip->is_signed)
val ^= 0x80;
timer_cnt = val * CUR_DIV() / 256;
if (timer_cnt && chip->enable) {
spin_lock(&i8253_lock);
spin_lock_irqsave(&i8253_lock, flags);
if (!nforce_wa) {
outb_p(chip->val61, 0x61);
outb_p(timer_cnt, 0x42);
@ -82,12 +79,39 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
outb(chip->val61 ^ 2, 0x61);
chip->thalf = 1;
}
spin_unlock(&i8253_lock);
spin_unlock_irqrestore(&i8253_lock, flags);
}
chip->ns_rem = PCSP_PERIOD_NS();
ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem);
chip->ns_rem -= ns;
return ns;
}
enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
{
struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
struct snd_pcm_substream *substream;
int periods_elapsed, pointer_update;
size_t period_bytes, buffer_bytes;
unsigned long ns;
unsigned long flags;
pointer_update = !chip->thalf;
ns = pcsp_timer_update(handle);
if (!ns)
return HRTIMER_NORESTART;
/* update the playback position */
substream = chip->playback_substream;
if (!substream)
return HRTIMER_NORESTART;
period_bytes = snd_pcm_lib_period_bytes(substream);
buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
chip->playback_ptr += PCSP_INDEX_INC() * fmt_size;
spin_lock_irqsave(&chip->substream_lock, flags);
chip->playback_ptr += PCSP_INDEX_INC() * chip->fmt_size;
periods_elapsed = chip->playback_ptr - chip->period_ptr;
if (periods_elapsed < 0) {
#if PCSP_DEBUG
@ -102,41 +126,30 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
* or ALSA will BUG on us. */
chip->playback_ptr %= buffer_bytes;
snd_pcm_stream_unlock(substream);
if (periods_elapsed) {
snd_pcm_period_elapsed(substream);
chip->period_ptr += periods_elapsed * period_bytes;
chip->period_ptr %= buffer_bytes;
}
spin_unlock_irqrestore(&chip->substream_lock, flags);
spin_unlock_irq(&chip->substream_lock);
if (periods_elapsed)
tasklet_schedule(&pcsp_pcm_tasklet);
if (!atomic_read(&chip->timer_active))
return HRTIMER_NORESTART;
hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
chip->ns_rem = PCSP_PERIOD_NS();
ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem);
chip->ns_rem -= ns;
hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
ktime_set(0, ns));
return HRTIMER_RESTART;
exit_nr_unlock2:
snd_pcm_stream_unlock(substream);
exit_nr_unlock1:
spin_unlock_irq(&chip->substream_lock);
return HRTIMER_NORESTART;
}
static void pcsp_start_playing(struct snd_pcsp *chip)
static int pcsp_start_playing(struct snd_pcsp *chip)
{
unsigned long ns;
#if PCSP_DEBUG
printk(KERN_INFO "PCSP: start_playing called\n");
#endif
if (atomic_read(&chip->timer_active)) {
printk(KERN_ERR "PCSP: Timer already active\n");
return;
return -EIO;
}
spin_lock(&i8253_lock);
@ -146,7 +159,12 @@ static void pcsp_start_playing(struct snd_pcsp *chip)
atomic_set(&chip->timer_active, 1);
chip->thalf = 0;
hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
ns = pcsp_timer_update(&pcsp_chip.timer);
if (!ns)
return -EIO;
hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL);
return 0;
}
static void pcsp_stop_playing(struct snd_pcsp *chip)
@ -165,26 +183,35 @@ static void pcsp_stop_playing(struct snd_pcsp *chip)
spin_unlock(&i8253_lock);
}
/*
* Force to stop and sync the stream
*/
void pcsp_sync_stop(struct snd_pcsp *chip)
{
local_irq_disable();
pcsp_stop_playing(chip);
local_irq_enable();
hrtimer_cancel(&chip->timer);
tasklet_kill(&pcsp_pcm_tasklet);
}
static int snd_pcsp_playback_close(struct snd_pcm_substream *substream)
{
struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
#if PCSP_DEBUG
printk(KERN_INFO "PCSP: close called\n");
#endif
if (atomic_read(&chip->timer_active)) {
printk(KERN_ERR "PCSP: timer still active\n");
pcsp_stop_playing(chip);
}
spin_lock_irq(&chip->substream_lock);
pcsp_sync_stop(chip);
chip->playback_substream = NULL;
spin_unlock_irq(&chip->substream_lock);
return 0;
}
static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
int err;
pcsp_sync_stop(chip);
err = snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
if (err < 0)
@ -194,9 +221,11 @@ static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream,
static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream)
{
struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
#if PCSP_DEBUG
printk(KERN_INFO "PCSP: hw_free called\n");
#endif
pcsp_sync_stop(chip);
return snd_pcm_lib_free_pages(substream);
}
@ -212,8 +241,12 @@ static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream)
snd_pcm_lib_period_bytes(substream),
substream->runtime->periods);
#endif
pcsp_sync_stop(chip);
chip->playback_ptr = 0;
chip->period_ptr = 0;
chip->fmt_size =
snd_pcm_format_physical_width(substream->runtime->format) >> 3;
chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
return 0;
}
@ -226,8 +259,7 @@ static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
pcsp_start_playing(chip);
break;
return pcsp_start_playing(chip);
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
pcsp_stop_playing(chip);
@ -242,7 +274,11 @@ static snd_pcm_uframes_t snd_pcsp_playback_pointer(struct snd_pcm_substream
*substream)
{
struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
return bytes_to_frames(substream->runtime, chip->playback_ptr);
unsigned int pos;
spin_lock(&chip->substream_lock);
pos = chip->playback_ptr;
spin_unlock(&chip->substream_lock);
return bytes_to_frames(substream->runtime, pos);
}
static struct snd_pcm_hardware snd_pcsp_playback = {
@ -279,9 +315,7 @@ static int snd_pcsp_playback_open(struct snd_pcm_substream *substream)
return -EBUSY;
}
runtime->hw = snd_pcsp_playback;
spin_lock_irq(&chip->substream_lock);
chip->playback_substream = substream;
spin_unlock_irq(&chip->substream_lock);
return 0;
}

2
sound/drivers/vx/vx_core.c
View File

@ -548,7 +548,7 @@ irqreturn_t snd_vx_irq_handler(int irq, void *dev)
(chip->chip_status & VX_STAT_IS_STALE))
return IRQ_NONE;
if (! vx_test_and_ack(chip))
tasklet_hi_schedule(&chip->tq);
tasklet_schedule(&chip->tq);
return IRQ_HANDLED;
}

2
sound/drivers/vx/vx_pcm.c
View File

@ -823,7 +823,7 @@ static int vx_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
* we trigger the pipe using tasklet, so that the interrupts are
* issued surely after the trigger is completed.
*/
tasklet_hi_schedule(&pipe->start_tq);
tasklet_schedule(&pipe->start_tq);
chip->pcm_running++;
pipe->running = 1;
break;

4
sound/isa/sb/sb8.c
View File

@ -140,8 +140,10 @@ static int __devinit snd_sb8_probe(struct device *pdev, unsigned int dev)
break;
}
}
if (i >= ARRAY_SIZE(possible_ports))
if (i >= ARRAY_SIZE(possible_ports)) {
err = -EINVAL;
goto _err;
}
}
acard->chip = chip;

127
sound/pci/Kconfig
View File

@ -208,7 +208,8 @@ config SND_OXYGEN
* AuzenTech X-Meridian
* Bgears b-Enspirer
* Club3D Theatron DTS
* HT-Omega Claro
* HT-Omega Claro (plus)
* HT-Omega Claro halo (XT)
* Razer Barracuda AC-1
* Sondigo Inferno
@ -497,129 +498,7 @@ config SND_FM801_TEA575X
depends on SND_FM801_TEA575X_BOOL
default SND_FM801
config SND_HDA_INTEL
tristate "Intel HD Audio"
select SND_PCM
select SND_VMASTER
help
Say Y here to include support for Intel "High Definition
Audio" (Azalia) motherboard devices.
To compile this driver as a module, choose M here: the module
will be called snd-hda-intel.
config SND_HDA_HWDEP
bool "Build hwdep interface for HD-audio driver"
depends on SND_HDA_INTEL
select SND_HWDEP
help
Say Y here to build a hwdep interface for HD-audio driver.
This interface can be used for out-of-band communication
with codecs for debugging purposes.
config SND_HDA_INPUT_BEEP
bool "Support digital beep via input layer"
depends on SND_HDA_INTEL
depends on INPUT=y || INPUT=SND_HDA_INTEL
help
Say Y here to build a digital beep interface for HD-audio
driver. This interface is used to generate digital beeps.
config SND_HDA_CODEC_REALTEK
bool "Build Realtek HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include Realtek HD-audio codec support in
snd-hda-intel driver, such as ALC880.
config SND_HDA_CODEC_ANALOG
bool "Build Analog Device HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include Analog Device HD-audio codec support in
snd-hda-intel driver, such as AD1986A.
config SND_HDA_CODEC_SIGMATEL
bool "Build IDT/Sigmatel HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include IDT (Sigmatel) HD-audio codec support in
snd-hda-intel driver, such as STAC9200.
config SND_HDA_CODEC_VIA
bool "Build VIA HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include VIA HD-audio codec support in
snd-hda-intel driver, such as VT1708.
config SND_HDA_CODEC_ATIHDMI
bool "Build ATI HDMI HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include ATI HDMI HD-audio codec support in
snd-hda-intel driver, such as ATI RS600 HDMI.
config SND_HDA_CODEC_NVHDMI
bool "Build NVIDIA HDMI HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include NVIDIA HDMI HD-audio codec support in
snd-hda-intel driver, such as NVIDIA MCP78 HDMI.
config SND_HDA_CODEC_CONEXANT
bool "Build Conexant HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include Conexant HD-audio codec support in
snd-hda-intel driver, such as CX20549.
config SND_HDA_CODEC_CMEDIA
bool "Build C-Media HD-audio codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include C-Media HD-audio codec support in
snd-hda-intel driver, such as CMI9880.
config SND_HDA_CODEC_SI3054
bool "Build Silicon Labs 3054 HD-modem codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include Silicon Labs 3054 HD-modem codec
(and compatibles) support in snd-hda-intel driver.
config SND_HDA_GENERIC
bool "Enable generic HD-audio codec parser"
depends on SND_HDA_INTEL
default y
help
Say Y here to enable the generic HD-audio codec parser
in snd-hda-intel driver.
config SND_HDA_POWER_SAVE
bool "Aggressive power-saving on HD-audio"
depends on SND_HDA_INTEL && EXPERIMENTAL
help
Say Y here to enable more aggressive power-saving mode on
HD-audio driver. The power-saving timeout can be configured
via power_save option or over sysfs on-the-fly.
config SND_HDA_POWER_SAVE_DEFAULT
int "Default time-out for HD-audio power-save mode"
depends on SND_HDA_POWER_SAVE
default 0
help
The default time-out value in seconds for HD-audio automatic
power-save mode. 0 means to disable the power-save mode.
source "sound/pci/hda/Kconfig"
config SND_HDSP
tristate "RME Hammerfall DSP Audio"

2
sound/pci/ac97/ac97_codec.c
View File

@ -175,7 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
{ 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL},
{ 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL},
{ 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL},
{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL},
{ 0x574d4C12, 0xffffffff, "WM9711,WM9712,WM9715", patch_wolfson11, NULL},
{ 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF},
{ 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL },
{ 0x594d4802, 0xffffffff, "YMF752", NULL, NULL },

7
sound/pci/ac97/ac97_patch.c
View File

@ -2054,8 +2054,9 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
.get = snd_ac97_ad1888_lohpsel_get,
.put = snd_ac97_ad1888_lohpsel_put
},
AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, 2, 1, 1),
AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, 12, 1, 1),
AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, AC97_AD_VREFD_SHIFT, 1, 1),
AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2,
AC97_AD_HPFD_SHIFT, 1, 1),
AC97_SINGLE("Spread Front to Surround and Center/LFE", AC97_AD_MISC, 7, 1, 0),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@ -2832,6 +2833,8 @@ static int patch_alc655(struct snd_ac97 * ac97)
val &= ~(1 << 1); /* Pin 47 is EAPD (for internal speaker) */
else
val |= (1 << 1); /* Pin 47 is spdif input pin */
/* this seems missing on some hardwares */
ac97->ext_id |= AC97_EI_SPDIF;
}
val &= ~(1 << 12); /* vref enable */
snd_ac97_write_cache(ac97, 0x7a, val);

30
sound/pci/ca0106/ca0106.h
View File

@ -664,10 +664,14 @@ struct snd_ca0106_pcm {
struct snd_ca0106_details {
u32 serial;
char * name;
int ac97;
int gpio_type;
int i2c_adc;
int spi_dac;
int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in.
ac97 = 1 -> Default to AC97 in. */
int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in
gpio_type = 2 -> shared side-out/line-in. */
int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume
controls, phone, mic, line-in and aux. */
int spi_dac; /* spi_dac=1 adds the mute switch for each analog
output, front, rear, etc. */
};
// definition of the chip-specific record
@ -686,11 +690,12 @@ struct snd_ca0106 {
spinlock_t emu_lock;
struct snd_ac97 *ac97;
struct snd_pcm *pcm;
struct snd_pcm *pcm[4];
struct snd_ca0106_channel playback_channels[4];
struct snd_ca0106_channel capture_channels[4];
u32 spdif_bits[4]; /* s/pdif out setup */
u32 spdif_bits[4]; /* s/pdif out default setup */
u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */
int spdif_enable;
int capture_source;
int i2c_capture_source;
@ -703,6 +708,11 @@ struct snd_ca0106 {
struct snd_ca_midi midi2;
u16 spi_dac_reg[16];
#ifdef CONFIG_PM
#define NUM_SAVED_VOLUMES 9
unsigned int saved_vol[NUM_SAVED_VOLUMES];
#endif
};
int snd_ca0106_mixer(struct snd_ca0106 *emu);
@ -721,3 +731,11 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
int snd_ca0106_spi_write(struct snd_ca0106 * emu,
unsigned int data);
#ifdef CONFIG_PM
void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip);
void snd_ca0106_mixer_resume(struct snd_ca0106 *chip);
#else
#define snd_ca0106_mixer_suspend(chip) do { } while (0)
#define snd_ca0106_mixer_resume(chip) do { } while (0)
#endif

552
sound/pci/ca0106/ca0106_main.c
View File

@ -254,7 +254,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
.name = "MSI K8N Diamond MB",
.gpio_type = 2,
.i2c_adc = 1,
.spi_dac = 2 } ,
.spi_dac = 1 } ,
/* Shuttle XPC SD31P which has an onboard Creative Labs
* Sound Blaster Live! 24-bit EAX
* high-definition 7.1 audio processor".
@ -305,9 +305,15 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
#if 0 /* FIXME: looks like 44.1kHz capture causes noisy output on 48kHz */
.rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
.rate_min = 44100,
#else
.rates = (SNDRV_PCM_RATE_48000 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
.rate_min = 48000,
#endif /* FIXME */
.rate_max = 192000,
.channels_min = 2,
.channels_max = 2,
@ -479,6 +485,15 @@ static const int spi_dacd_bit[] = {
[PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT,
};
static void restore_spdif_bits(struct snd_ca0106 *chip, int idx)
{
if (chip->spdif_str_bits[idx] != chip->spdif_bits[idx]) {
chip->spdif_str_bits[idx] = chip->spdif_bits[idx];
snd_ca0106_ptr_write(chip, SPCS0 + idx, 0,
chip->spdif_str_bits[idx]);
}
}
/* open_playback callback */
static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream,
int channel_id)
@ -524,6 +539,9 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr
if (err < 0)
return err;
}
restore_spdif_bits(chip, channel_id);
return 0;
}
@ -535,6 +553,8 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream)
struct snd_ca0106_pcm *epcm = runtime->private_data;
chip->playback_channels[epcm->channel_id].use = 0;
restore_spdif_bits(chip, epcm->channel_id);
if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) {
const int reg = spi_dacd_reg[epcm->channel_id];
@ -847,15 +867,18 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
struct snd_pcm_substream *s;
u32 basic = 0;
u32 extended = 0;
int running=0;
u32 bits;
int running = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
running=1;
case SNDRV_PCM_TRIGGER_RESUME:
running = 1;
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
default:
running=0;
running = 0;
break;
}
snd_pcm_group_for_each_entry(s, substream) {
@ -865,22 +888,32 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
runtime = s->runtime;
epcm = runtime->private_data;
channel = epcm->channel_id;
//snd_printk("channel=%d\n",channel);
/* snd_printk("channel=%d\n",channel); */
epcm->running = running;
basic |= (0x1<<channel);
extended |= (0x10<<channel);
basic |= (0x1 << channel);
extended |= (0x10 << channel);
snd_pcm_trigger_done(s, substream);
}
//snd_printk("basic=0x%x, extended=0x%x\n",basic, extended);
/* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (extended));
snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(basic));
case SNDRV_PCM_TRIGGER_RESUME:
bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
bits |= extended;
snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
bits |= basic;
snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
break;
case SNDRV_PCM_TRIGGER_STOP:
snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(basic));
snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(extended));
case SNDRV_PCM_TRIGGER_SUSPEND:
bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
bits &= ~basic;
snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
bits &= ~extended;
snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
break;
default:
result = -EINVAL;
@ -1103,21 +1136,13 @@ static int snd_ca0106_ac97(struct snd_ca0106 *chip)
return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
}
static void ca0106_stop_chip(struct snd_ca0106 *chip);
static int snd_ca0106_free(struct snd_ca0106 *chip)
{
if (chip->res_port != NULL) { /* avoid access to already used hardware */
// disable interrupts
snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
outl(0, chip->port + INTE);
snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
udelay(1000);
// disable audio
//outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
outl(0, chip->port + HCFG);
/* FIXME: We need to stop and DMA transfers here.
* But as I am not sure how yet, we cannot from the dma pages.
* So we can fix: snd-malloc: Memory leak? pages not freed = 8
*/
if (chip->res_port != NULL) {
/* avoid access to already used hardware */
ca0106_stop_chip(chip);
}
if (chip->irq >= 0)
free_irq(chip->irq, chip);
@ -1203,15 +1228,14 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
return IRQ_HANDLED;
}
static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct snd_pcm **rpcm)
static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device)
{
struct snd_pcm *pcm;
struct snd_pcm_substream *substream;
int err;
if (rpcm)
*rpcm = NULL;
if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0)
err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm);
if (err < 0)
return err;
pcm->private_data = emu;
@ -1238,7 +1262,6 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s
pcm->info_flags = 0;
pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
strcpy(pcm->name, "CA0106");
emu->pcm = pcm;
for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
substream;
@ -1260,8 +1283,7 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s
return err;
}
if (rpcm)
*rpcm = pcm;
emu->pcm[device] = pcm;
return 0;
}
@ -1301,89 +1323,10 @@ static unsigned int i2c_adc_init[][2] = {
{ 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */
};
static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
struct pci_dev *pci,
struct snd_ca0106 **rchip)
static void ca0106_init_chip(struct snd_ca0106 *chip, int resume)
{
struct snd_ca0106 *chip;
struct snd_ca0106_details *c;
int err;
int ch;
static struct snd_device_ops ops = {
.dev_free = snd_ca0106_dev_free,
};
*rchip = NULL;
if ((err = pci_enable_device(pci)) < 0)
return err;
if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
printk(KERN_ERR "error to set 32bit mask DMA\n");
pci_disable_device(pci);
return -ENXIO;
}
chip = kzalloc(sizeof(*chip), GFP_KERNEL);
if (chip == NULL) {
pci_disable_device(pci);
return -ENOMEM;
}
chip->card = card;
chip->pci = pci;
chip->irq = -1;
spin_lock_init(&chip->emu_lock);
chip->port = pci_resource_start(pci, 0);
if ((chip->res_port = request_region(chip->port, 0x20,
"snd_ca0106")) == NULL) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot allocate the port\n");
return -EBUSY;
}
if (request_irq(pci->irq, snd_ca0106_interrupt,
IRQF_SHARED, "snd_ca0106", chip)) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
}
chip->irq = pci->irq;
/* This stores the periods table. */
if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) {
snd_ca0106_free(chip);
return -ENOMEM;
}
pci_set_master(pci);
/* read serial */
pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
#if 1
printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model,
pci->revision, chip->serial);
#endif
strcpy(card->driver, "CA0106");
strcpy(card->shortname, "CA0106");
for (c = ca0106_chip_details; c->serial; c++) {
if (subsystem[dev]) {
if (c->serial == subsystem[dev])
break;
} else if (c->serial == chip->serial)
break;
}
chip->details = c;
if (subsystem[dev]) {
printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n",
c->name, chip->serial, subsystem[dev]);
}
sprintf(card->longname, "%s at 0x%lx irq %i",
c->name, chip->port, chip->irq);
unsigned int def_bits;
outl(0, chip->port + INTE);
@ -1401,31 +1344,22 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
* AN = 0 (Audio data)
* P = 0 (Consumer)
*/
snd_ca0106_ptr_write(chip, SPCS0, 0,
chip->spdif_bits[0] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
def_bits =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT;
if (!resume) {
chip->spdif_str_bits[0] = chip->spdif_bits[0] = def_bits;
chip->spdif_str_bits[1] = chip->spdif_bits[1] = def_bits;
chip->spdif_str_bits[2] = chip->spdif_bits[2] = def_bits;
chip->spdif_str_bits[3] = chip->spdif_bits[3] = def_bits;
}
/* Only SPCS1 has been tested */
snd_ca0106_ptr_write(chip, SPCS1, 0,
chip->spdif_bits[1] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
snd_ca0106_ptr_write(chip, SPCS2, 0,
chip->spdif_bits[2] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
snd_ca0106_ptr_write(chip, SPCS3, 0,
chip->spdif_bits[3] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
snd_ca0106_ptr_write(chip, SPCS1, 0, chip->spdif_str_bits[1]);
snd_ca0106_ptr_write(chip, SPCS0, 0, chip->spdif_str_bits[0]);
snd_ca0106_ptr_write(chip, SPCS2, 0, chip->spdif_str_bits[2]);
snd_ca0106_ptr_write(chip, SPCS3, 0, chip->spdif_str_bits[3]);
snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000);
snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000);
@ -1433,92 +1367,124 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
/* Write 0x8000 to AC97_REC_GAIN to mute it. */
outb(AC97_REC_GAIN, chip->port + AC97ADDRESS);
outw(0x8000, chip->port + AC97DATA);
#if 0
#if 0 /* FIXME: what are these? */
snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006);
snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006);
snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006);
snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006);
#endif
//snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */
/* OSS drivers set this. */
/* snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); */
/* Analog or Digital output */
snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. Use 0x000f0000 for surround71 */
chip->spdif_enable = 0; /* Set digital SPDIF output off */
//snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
//snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */
/* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers.
* Use 0x000f0000 for surround71
*/
snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000);
chip->spdif_enable = 0; /* Set digital SPDIF output off */
/*snd_ca0106_ptr_write(chip, 0x45, 0, 0);*/ /* Analogue out */
/*snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00);*/ /* Digital out */
/* goes to 0x40c80000 when doing SPDIF IN/OUT */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000);
/* (Mute) CAPTURE feedback into PLAYBACK volume.
* Only lower 16 bits matter.
*/
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff);
/* SPDIF IN Volume */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000);
/* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000);
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410);
snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676);
snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410);
snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676);
for(ch = 0; ch < 4; ch++) {
snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */
for (ch = 0; ch < 4; ch++) {
/* Only high 16 bits matter */
snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030);
snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
//snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */
//snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */
#if 0 /* Mute */
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040);
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040);
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff);
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff);
#endif
}
if (chip->details->i2c_adc == 1) {
/* Select MIC, Line in, TAD in, AUX in */
snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
/* Default to CAPTURE_SOURCE to i2s in */
chip->capture_source = 3;
if (!resume)
chip->capture_source = 3;
} else if (chip->details->ac97 == 1) {
/* Default to AC97 in */
snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4);
/* Default to CAPTURE_SOURCE to AC97 in */
chip->capture_source = 4;
if (!resume)
chip->capture_source = 4;
} else {
/* Select MIC, Line in, TAD in, AUX in */
snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
/* Default to Set CAPTURE_SOURCE to i2s in */
chip->capture_source = 3;
if (!resume)
chip->capture_source = 3;
}
if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */
/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
if (chip->details->gpio_type == 2) {
/* The SB0438 use GPIO differently. */
/* FIXME: Still need to find out what the other GPIO bits do.
* E.g. For digital spdif out.
*/
outl(0x0, chip->port+GPIO);
//outl(0x00f0e000, chip->port+GPIO); /* Analog */
/* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */
outl(0x005f5301, chip->port+GPIO); /* Analog */
} else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */
/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
} else if (chip->details->gpio_type == 1) {
/* The SB0410 and SB0413 use GPIO differently. */
/* FIXME: Still need to find out what the other GPIO bits do.
* E.g. For digital spdif out.
*/
outl(0x0, chip->port+GPIO);
//outl(0x00f0e000, chip->port+GPIO); /* Analog */
/* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */
outl(0x005f5301, chip->port+GPIO); /* Analog */
} else {
outl(0x0, chip->port+GPIO);
outl(0x005f03a3, chip->port+GPIO); /* Analog */
//outl(0x005f02a2, chip->port+GPIO); /* SPDIF */
/* outl(0x005f02a2, chip->port+GPIO); */ /* SPDIF */
}
snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
//outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
//outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
//outl(0x00000009, chip->port+HCFG);
outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */
/* outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); */
/* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
/* outl(0x00001409, chip->port+HCFG); */
/* outl(0x00000009, chip->port+HCFG); */
/* AC97 2.0, Enable outputs. */
outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG);
if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
if (chip->details->i2c_adc == 1) {
/* The SB0410 and SB0413 use I2C to control ADC. */
int size, n;
size = ARRAY_SIZE(i2c_adc_init);
//snd_printk("I2C:array size=0x%x\n", size);
for (n=0; n < size; n++) {
snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]);
/* snd_printk("I2C:array size=0x%x\n", size); */
for (n = 0; n < size; n++)
snd_ca0106_i2c_write(chip, i2c_adc_init[n][0],
i2c_adc_init[n][1]);
for (n = 0; n < 4; n++) {
chip->i2c_capture_volume[n][0] = 0xcf;
chip->i2c_capture_volume[n][1] = 0xcf;
}
for (n=0; n < 4; n++) {
chip->i2c_capture_volume[n][0]= 0xcf;
chip->i2c_capture_volume[n][1]= 0xcf;
}
chip->i2c_capture_source=2; /* Line in */
//snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */
chip->i2c_capture_source = 2; /* Line in */
/* Enable Line-in capture. MIC in currently untested. */
/* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); */
}
if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */
if (chip->details->spi_dac == 1) {
/* The SB0570 use SPI to control DAC. */
int size, n;
size = ARRAY_SIZE(spi_dac_init);
@ -1530,9 +1496,112 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
chip->spi_dac_reg[reg] = spi_dac_init[n];
}
}
}
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
chip, &ops)) < 0) {
static void ca0106_stop_chip(struct snd_ca0106 *chip)
{
/* disable interrupts */
snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
outl(0, chip->port + INTE);
snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
udelay(1000);
/* disable audio */
/* outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); */
outl(0, chip->port + HCFG);
/* FIXME: We need to stop and DMA transfers here.
* But as I am not sure how yet, we cannot from the dma pages.
* So we can fix: snd-malloc: Memory leak? pages not freed = 8
*/
}
static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
struct pci_dev *pci,
struct snd_ca0106 **rchip)
{
struct snd_ca0106 *chip;
struct snd_ca0106_details *c;
int err;
static struct snd_device_ops ops = {
.dev_free = snd_ca0106_dev_free,
};
*rchip = NULL;
err = pci_enable_device(pci);
if (err < 0)
return err;
if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
printk(KERN_ERR "error to set 32bit mask DMA\n");
pci_disable_device(pci);
return -ENXIO;
}
chip = kzalloc(sizeof(*chip), GFP_KERNEL);
if (chip == NULL) {
pci_disable_device(pci);
return -ENOMEM;
}
chip->card = card;
chip->pci = pci;
chip->irq = -1;
spin_lock_init(&chip->emu_lock);
chip->port = pci_resource_start(pci, 0);
chip->res_port = request_region(chip->port, 0x20, "snd_ca0106");
if (!chip->res_port) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot allocate the port\n");
return -EBUSY;
}
if (request_irq(pci->irq, snd_ca0106_interrupt,
IRQF_SHARED, "snd_ca0106", chip)) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
}
chip->irq = pci->irq;
/* This stores the periods table. */
if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
1024, &chip->buffer) < 0) {
snd_ca0106_free(chip);
return -ENOMEM;
}
pci_set_master(pci);
/* read serial */
pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n",
chip->model, pci->revision, chip->serial);
strcpy(card->driver, "CA0106");
strcpy(card->shortname, "CA0106");
for (c = ca0106_chip_details; c->serial; c++) {
if (subsystem[dev]) {
if (c->serial == subsystem[dev])
break;
} else if (c->serial == chip->serial)
break;
}
chip->details = c;
if (subsystem[dev]) {
printk(KERN_INFO "snd-ca0106: Sound card name=%s, "
"subsystem=0x%x. Forced to subsystem=0x%x\n",
c->name, chip->serial, subsystem[dev]);
}
sprintf(card->longname, "%s at 0x%lx irq %i",
c->name, chip->port, chip->irq);
ca0106_init_chip(chip, 0);
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
snd_ca0106_free(chip);
return err;
}
@ -1629,7 +1698,7 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
static int dev;
struct snd_card *card;
struct snd_ca0106 *chip;
int err;
int i, err;
if (dev >= SNDRV_CARDS)
return -ENODEV;
@ -1642,44 +1711,31 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
if (card == NULL)
return -ENOMEM;
if ((err = snd_ca0106_create(dev, card, pci, &chip)) < 0) {
snd_card_free(card);
return err;
err = snd_ca0106_create(dev, card, pci, &chip);
if (err < 0)
goto error;
card->private_data = chip;
for (i = 0; i < 4; i++) {
err = snd_ca0106_pcm(chip, i);
if (err < 0)
goto error;
}
if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) {
snd_card_free(card);
return err;
}
if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) {
snd_card_free(card);
return err;
}
if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) {
snd_card_free(card);
return err;
}
if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) {
snd_card_free(card);
return err;
}
if (chip->details->ac97 == 1) { /* The SB0410 and SB0413 do not have an AC97 chip. */
if ((err = snd_ca0106_ac97(chip)) < 0) {
snd_card_free(card);
return err;
}
}
if ((err = snd_ca0106_mixer(chip)) < 0) {
snd_card_free(card);
return err;
if (chip->details->ac97 == 1) {
/* The SB0410 and SB0413 do not have an AC97 chip. */
err = snd_ca0106_ac97(chip);
if (err < 0)
goto error;
}
err = snd_ca0106_mixer(chip);
if (err < 0)
goto error;
snd_printdd("ca0106: probe for MIDI channel A ...");
if ((err = snd_ca0106_midi(chip,CA0106_MIDI_CHAN_A)) < 0) {
snd_card_free(card);
snd_printdd(" failed, err=0x%x\n",err);
return err;
}
err = snd_ca0106_midi(chip, CA0106_MIDI_CHAN_A);
if (err < 0)
goto error;
snd_printdd(" done.\n");
#ifdef CONFIG_PROC_FS
@ -1688,14 +1744,17 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
snd_card_set_dev(card, &pci->dev);
if ((err = snd_card_register(card)) < 0) {
snd_card_free(card);
return err;
}
err = snd_card_register(card);
if (err < 0)
goto error;
pci_set_drvdata(pci, card);
dev++;
return 0;
error:
snd_card_free(card);
return err;
}
static void __devexit snd_ca0106_remove(struct pci_dev *pci)
@ -1704,6 +1763,59 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
#ifdef CONFIG_PM
static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state)
{
struct snd_card *card = pci_get_drvdata(pci);
struct snd_ca0106 *chip = card->private_data;
int i;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
for (i = 0; i < 4; i++)
snd_pcm_suspend_all(chip->pcm[i]);
if (chip->details->ac97)
snd_ac97_suspend(chip->ac97);
snd_ca0106_mixer_suspend(chip);
ca0106_stop_chip(chip);
pci_disable_device(pci);
pci_save_state(pci);
pci_set_power_state(pci, pci_choose_state(pci, state));
return 0;
}
static int snd_ca0106_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct snd_ca0106 *chip = card->private_data;
int i;
pci_set_power_state(pci, PCI_D0);
pci_restore_state(pci);
if (pci_enable_device(pci) < 0) {
snd_card_disconnect(card);
return -EIO;
}
pci_set_master(pci);
ca0106_init_chip(chip, 1);
if (chip->details->ac97)
snd_ac97_resume(chip->ac97);
snd_ca0106_mixer_resume(chip);
if (chip->details->spi_dac) {
for (i = 0; i < ARRAY_SIZE(chip->spi_dac_reg); i++)
snd_ca0106_spi_write(chip, chip->spi_dac_reg[i]);
}
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
}
#endif
// PCI IDs
static struct pci_device_id snd_ca0106_ids[] = {
{ 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */
@ -1717,6 +1829,10 @@ static struct pci_driver driver = {
.id_table = snd_ca0106_ids,
.probe = snd_ca0106_probe,
.remove = __devexit_p(snd_ca0106_remove),
#ifdef CONFIG_PM
.suspend = snd_ca0106_suspend,
.resume = snd_ca0106_resume,
#endif
};
// initialization of the module

263
sound/pci/ca0106/ca0106_mixer.c
View File

@ -75,6 +75,84 @@
#include "ca0106.h"
static void ca0106_spdif_enable(struct snd_ca0106 *emu)
{
unsigned int val;
if (emu->spdif_enable) {
/* Digital */
snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000;
snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
val = inl(emu->port + GPIO) & ~0x101;
outl(val, emu->port + GPIO);
} else {
/* Analog */
snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000;
snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
val = inl(emu->port + GPIO) | 0x101;
outl(val, emu->port + GPIO);
}
}
static void ca0106_set_capture_source(struct snd_ca0106 *emu)
{
unsigned int val = emu->capture_source;
unsigned int source, mask;
source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
}
static void ca0106_set_i2c_capture_source(struct snd_ca0106 *emu,
unsigned int val, int force)
{
unsigned int ngain, ogain;
u32 source;
snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
ngain = emu->i2c_capture_volume[val][0]; /* Left */
ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
if (force || ngain != ogain)
snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ngain & 0xff);
ngain = emu->i2c_capture_volume[val][1]; /* Right */
ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Right */
if (force || ngain != ogain)
snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ngain & 0xff);
source = 1 << val;
snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
emu->i2c_capture_source = val;
}
static void ca0106_set_capture_mic_line_in(struct snd_ca0106 *emu)
{
u32 tmp;
if (emu->capture_mic_line_in) {
/* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
tmp = inl(emu->port+GPIO) & ~0x400;
tmp = tmp | 0x400;
outl(tmp, emu->port+GPIO);
/* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */
} else {
/* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
tmp = inl(emu->port+GPIO) & ~0x400;
outl(tmp, emu->port+GPIO);
/* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */
}
}
static void ca0106_set_spdif_bits(struct snd_ca0106 *emu, int idx)
{
snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, emu->spdif_str_bits[idx]);
}
/*
*/
static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
@ -95,30 +173,12 @@ static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol,
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
int change = 0;
u32 mask;
val = !!ucontrol->value.integer.value[0];
change = (emu->spdif_enable != val);
if (change) {
emu->spdif_enable = val;
if (val) {
/* Digital */
snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000);
mask = inl(emu->port + GPIO) & ~0x101;
outl(mask, emu->port + GPIO);
} else {
/* Analog */
snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000);
mask = inl(emu->port + GPIO) | 0x101;
outl(mask, emu->port + GPIO);
}
ca0106_spdif_enable(emu);
}
return change;
}
@ -154,8 +214,6 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
int change = 0;
u32 mask;
u32 source;
val = ucontrol->value.enumerated.item[0] ;
if (val >= 6)
@ -163,9 +221,7 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
change = (emu->capture_source != val);
if (change) {
emu->capture_source = val;
source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
ca0106_set_capture_source(emu);
}
return change;
}
@ -200,9 +256,7 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
{
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int source_id;
unsigned int ngain, ogain;
int change = 0;
u32 source;
/* If the capture source has changed,
* update the capture volume from the cached value
* for the particular source.
@ -212,18 +266,7 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
return -EINVAL;
change = (emu->i2c_capture_source != source_id);
if (change) {
snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
ngain = emu->i2c_capture_volume[source_id][0]; /* Left */
ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
if (ngain != ogain)
snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff));
ngain = emu->i2c_capture_volume[source_id][1]; /* Left */
ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Left */
if (ngain != ogain)
snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
source = 1 << source_id;
snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
emu->i2c_capture_source = source_id;
ca0106_set_i2c_capture_source(emu, source_id, 0);
}
return change;
}
@ -271,7 +314,6 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
int change = 0;
u32 tmp;
val = ucontrol->value.enumerated.item[0] ;
if (val > 1)
@ -279,18 +321,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
change = (emu->capture_mic_line_in != val);
if (change) {
emu->capture_mic_line_in = val;
if (val) {
//snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
tmp = inl(emu->port+GPIO) & ~0x400;
tmp = tmp | 0x400;
outl(tmp, emu->port+GPIO);
//snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC);
} else {
//snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
tmp = inl(emu->port+GPIO) & ~0x400;
outl(tmp, emu->port+GPIO);
//snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN);
}
ca0106_set_capture_mic_line_in(emu);
}
return change;
}
@ -322,16 +353,33 @@ static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol,
return 0;
}
static int snd_ca0106_spdif_get(struct snd_kcontrol *kcontrol,
static void decode_spdif_bits(unsigned char *status, unsigned int bits)
{
status[0] = (bits >> 0) & 0xff;
status[1] = (bits >> 8) & 0xff;
status[2] = (bits >> 16) & 0xff;
status[3] = (bits >> 24) & 0xff;
}
static int snd_ca0106_spdif_get_default(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff;
ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff;
decode_spdif_bits(ucontrol->value.iec958.status,
emu->spdif_bits[idx]);
return 0;
}
static int snd_ca0106_spdif_get_stream(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
decode_spdif_bits(ucontrol->value.iec958.status,
emu->spdif_str_bits[idx]);
return 0;
}
@ -345,24 +393,48 @@ static int snd_ca0106_spdif_get_mask(struct snd_kcontrol *kcontrol,
return 0;
}
static int snd_ca0106_spdif_put(struct snd_kcontrol *kcontrol,
static unsigned int encode_spdif_bits(unsigned char *status)
{
return ((unsigned int)status[0] << 0) |
((unsigned int)status[1] << 8) |
((unsigned int)status[2] << 16) |
((unsigned int)status[3] << 24);
}
static int snd_ca0106_spdif_put_default(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
int change;
unsigned int val;
val = (ucontrol->value.iec958.status[0] << 0) |
(ucontrol->value.iec958.status[1] << 8) |
(ucontrol->value.iec958.status[2] << 16) |
(ucontrol->value.iec958.status[3] << 24);
change = val != emu->spdif_bits[idx];
if (change) {
snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, val);
val = encode_spdif_bits(ucontrol->value.iec958.status);
if (val != emu->spdif_bits[idx]) {
emu->spdif_bits[idx] = val;
/* FIXME: this isn't safe, but needed to keep the compatibility
* with older alsa-lib config
*/
emu->spdif_str_bits[idx] = val;
ca0106_set_spdif_bits(emu, idx);
return 1;
}
return change;
return 0;
}
static int snd_ca0106_spdif_put_stream(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
unsigned int val;
val = encode_spdif_bits(ucontrol->value.iec958.status);
if (val != emu->spdif_str_bits[idx]) {
emu->spdif_str_bits[idx] = val;
ca0106_set_spdif_bits(emu, idx);
return 1;
}
return 0;
}
static int snd_ca0106_volume_info(struct snd_kcontrol *kcontrol,
@ -573,8 +645,16 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
.count = 4,
.info = snd_ca0106_spdif_info,
.get = snd_ca0106_spdif_get,
.put = snd_ca0106_spdif_put
.get = snd_ca0106_spdif_get_default,
.put = snd_ca0106_spdif_put_default
},
{
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
.count = 4,
.info = snd_ca0106_spdif_info,
.get = snd_ca0106_spdif_get_stream,
.put = snd_ca0106_spdif_put_stream
},
};
@ -773,3 +853,50 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
return 0;
}
#ifdef CONFIG_PM
struct ca0106_vol_tbl {
unsigned int channel_id;
unsigned int reg;
};
static struct ca0106_vol_tbl saved_volumes[NUM_SAVED_VOLUMES] = {
{ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2 },
{ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2 },
{ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2 },
{ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2 },
{ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1 },
{ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1 },
{ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1 },
{ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1 },
{ 1, CAPTURE_CONTROL },
};
void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip)
{
int i;
/* save volumes */
for (i = 0; i < NUM_SAVED_VOLUMES; i++)
chip->saved_vol[i] =
snd_ca0106_ptr_read(chip, saved_volumes[i].reg,
saved_volumes[i].channel_id);
}
void snd_ca0106_mixer_resume(struct snd_ca0106 *chip)
{
int i;
for (i = 0; i < NUM_SAVED_VOLUMES; i++)
snd_ca0106_ptr_write(chip, saved_volumes[i].reg,
saved_volumes[i].channel_id,
chip->saved_vol[i]);
ca0106_spdif_enable(chip);
ca0106_set_capture_source(chip);
ca0106_set_i2c_capture_source(chip, chip->i2c_capture_source, 1);
for (i = 0; i < 4; i++)
ca0106_set_spdif_bits(chip, i);
if (chip->details->i2c_adc)
ca0106_set_capture_mic_line_in(chip);
}
#endif /* CONFIG_PM */

5
sound/pci/cs46xx/cs46xx_lib.c
View File

@ -3640,7 +3640,10 @@ int snd_cs46xx_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct snd_cs46xx *chip = card->private_data;
int i, amp_saved;
int amp_saved;
#ifdef CONFIG_SND_CS46XX_NEW_DSP
int i;
#endif
pci_set_power_state(pci, PCI_D0);
pci_restore_state(pci);

3
sound/pci/cs5535audio/Makefile
View File

@ -4,6 +4,9 @@
snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
ifdef CONFIG_MGEODE_LX
snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.o
endif
# Toplevel Module Dependency
obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o

12
sound/pci/cs5535audio/cs5535audio.c
View File

@ -159,10 +159,14 @@ static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au)
return err;
memset(&ac97, 0, sizeof(ac97));
ac97.scaps = AC97_SCAP_AUDIO|AC97_SCAP_SKIP_MODEM;
ac97.scaps = AC97_SCAP_AUDIO | AC97_SCAP_SKIP_MODEM
| AC97_SCAP_POWER_SAVE;
ac97.private_data = cs5535au;
ac97.pci = cs5535au->pci;
/* set any OLPC-specific scaps */
olpc_prequirks(card, &ac97);
if ((err = snd_ac97_mixer(pbus, &ac97, &cs5535au->ac97)) < 0) {
snd_printk(KERN_ERR "mixer failed\n");
return err;
@ -170,6 +174,12 @@ static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au)
snd_ac97_tune_hardware(cs5535au->ac97, ac97_quirks, ac97_quirk);
err = olpc_quirks(card, cs5535au->ac97);
if (err < 0) {
snd_printk(KERN_ERR "olpc quirks failed\n");
return err;
}
return 0;
}

39
sound/pci/cs5535audio/cs5535audio.h
View File

@ -78,6 +78,7 @@ struct cs5535audio_dma {
unsigned int buf_addr, buf_bytes;
unsigned int period_bytes, periods;
u32 saved_prd;
int pcm_open_flag;
};
struct cs5535audio {
@ -93,8 +94,46 @@ struct cs5535audio {
struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS];
};
#ifdef CONFIG_PM
int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
int snd_cs5535audio_resume(struct pci_dev *pci);
#endif
#if defined(CONFIG_OLPC) && defined(CONFIG_MGEODE_LX)
void __devinit olpc_prequirks(struct snd_card *card,
struct snd_ac97_template *ac97);
int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97);
void olpc_analog_input(struct snd_ac97 *ac97, int on);
void olpc_mic_bias(struct snd_ac97 *ac97, int on);
static inline void olpc_capture_open(struct snd_ac97 *ac97)
{
/* default to Analog Input off */
olpc_analog_input(ac97, 0);
/* enable MIC Bias for recording */
olpc_mic_bias(ac97, 1);
}
static inline void olpc_capture_close(struct snd_ac97 *ac97)
{
/* disable Analog Input */
olpc_analog_input(ac97, 0);
/* disable the MIC Bias (so the recording LED turns off) */
olpc_mic_bias(ac97, 0);
}
#else
static inline void olpc_prequirks(struct snd_card *card,
struct snd_ac97_template *ac97) { }
static inline int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
{
return 0;
}
static inline void olpc_analog_input(struct snd_ac97 *ac97, int on) { }
static inline void olpc_mic_bias(struct snd_ac97 *ac97, int on) { }
static inline void olpc_capture_open(struct snd_ac97 *ac97) { }
static inline void olpc_capture_close(struct snd_ac97 *ac97) { }
#endif
int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535audio);
#endif /* __SOUND_CS5535AUDIO_H */

179
sound/pci/cs5535audio/cs5535audio_olpc.c Normal file
View File

@ -0,0 +1,179 @@
/*
* OLPC XO-1 additional sound features
*
* Copyright © 2006 Jaya Kumar <jayakumar.lkml@gmail.com>
* Copyright © 2007-2008 Andres Salomon <dilinger@debian.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*/
#include <sound/core.h>
#include <sound/info.h>
#include <sound/control.h>
#include <sound/ac97_codec.h>
#include <asm/olpc.h>
#include "cs5535audio.h"
/*
* OLPC has an additional feature on top of the regular AD1888 codec features.
* It has an Analog Input mode that is switched into (after disabling the
* High Pass Filter) via GPIO. It is supported on B2 and later models.
*/
void olpc_analog_input(struct snd_ac97 *ac97, int on)
{
int err;
if (!machine_is_olpc())
return;
/* update the High Pass Filter (via AC97_AD_TEST2) */
err = snd_ac97_update_bits(ac97, AC97_AD_TEST2,
1 << AC97_AD_HPFD_SHIFT, on << AC97_AD_HPFD_SHIFT);
if (err < 0) {
snd_printk(KERN_ERR "setting High Pass Filter - %d\n", err);
return;
}
/* set Analog Input through GPIO */
if (on)
geode_gpio_set(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
else
geode_gpio_clear(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
}
/*
* OLPC XO-1's V_REFOUT is a mic bias enable.
*/
void olpc_mic_bias(struct snd_ac97 *ac97, int on)
{
int err;
if (!machine_is_olpc())
return;
on = on ? 0 : 1;
err = snd_ac97_update_bits(ac97, AC97_AD_MISC,
1 << AC97_AD_VREFD_SHIFT, on << AC97_AD_VREFD_SHIFT);
if (err < 0)
snd_printk(KERN_ERR "setting MIC Bias - %d\n", err);
}
static int olpc_dc_info(struct snd_kcontrol *kctl,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int olpc_dc_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
{
v->value.integer.value[0] = geode_gpio_isset(OLPC_GPIO_MIC_AC,
GPIO_OUTPUT_VAL);
return 0;
}
static int olpc_dc_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
{
struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl);
olpc_analog_input(cs5535au->ac97, v->value.integer.value[0]);
return 1;
}
static int olpc_mic_info(struct snd_kcontrol *kctl,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int olpc_mic_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
{
struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl);
struct snd_ac97 *ac97 = cs5535au->ac97;
int i;
i = (snd_ac97_read(ac97, AC97_AD_MISC) >> AC97_AD_VREFD_SHIFT) & 0x1;
v->value.integer.value[0] = i ? 0 : 1;
return 0;
}
static int olpc_mic_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
{
struct cs5535audio *cs5535au = snd_kcontrol_chip(kctl);
olpc_mic_bias(cs5535au->ac97, v->value.integer.value[0]);
return 1;
}
static struct snd_kcontrol_new olpc_cs5535audio_ctls[] __devinitdata = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "DC Mode Enable",
.info = olpc_dc_info,
.get = olpc_dc_get,
.put = olpc_dc_put,
.private_value = 0,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "MIC Bias Enable",
.info = olpc_mic_info,
.get = olpc_mic_get,
.put = olpc_mic_put,
.private_value = 0,
},
};
void __devinit olpc_prequirks(struct snd_card *card,
struct snd_ac97_template *ac97)
{
if (!machine_is_olpc())
return;
/* invert EAPD if on an OLPC B3 or higher */
if (olpc_board_at_least(olpc_board_pre(0xb3)))
ac97->scaps |= AC97_SCAP_INV_EAPD;
}
int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
{
struct snd_ctl_elem_id elem;
int i, err;
if (!machine_is_olpc())
return 0;
/* drop the original AD1888 HPF control */
memset(&elem, 0, sizeof(elem));
elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
strncpy(elem.name, "High Pass Filter Enable", sizeof(elem.name));
snd_ctl_remove_id(card, &elem);
/* drop the original V_REFOUT control */
memset(&elem, 0, sizeof(elem));
elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
strncpy(elem.name, "V_REFOUT Enable", sizeof(elem.name));
snd_ctl_remove_id(card, &elem);
/* add the OLPC-specific controls */
for (i = 0; i < ARRAY_SIZE(olpc_cs5535audio_ctls); i++) {
err = snd_ctl_add(card, snd_ctl_new1(&olpc_cs5535audio_ctls[i],
ac97->private_data));
if (err < 0)
return err;
}
/* turn off the mic by default */
olpc_mic_bias(ac97, 0);
return 0;
}

15
sound/pci/cs5535audio/cs5535audio_pcm.c
View File

@ -260,6 +260,9 @@ static int snd_cs5535audio_hw_params(struct snd_pcm_substream *substream,
err = cs5535audio_build_dma_packets(cs5535au, dma, substream,
params_periods(hw_params),
params_period_bytes(hw_params));
if (!err)
dma->pcm_open_flag = 1;
return err;
}
@ -268,6 +271,15 @@ static int snd_cs5535audio_hw_free(struct snd_pcm_substream *substream)
struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
struct cs5535audio_dma *dma = substream->runtime->private_data;
if (dma->pcm_open_flag) {
if (substream == cs5535au->playback_substream)
snd_ac97_update_power(cs5535au->ac97,
AC97_PCM_FRONT_DAC_RATE, 0);
else
snd_ac97_update_power(cs5535au->ac97,
AC97_PCM_LR_ADC_RATE, 0);
dma->pcm_open_flag = 0;
}
cs5535audio_clear_dma_packets(cs5535au, dma, substream);
return snd_pcm_lib_free_pages(substream);
}
@ -351,11 +363,14 @@ static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream)
if ((err = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
olpc_capture_open(cs5535au->ac97);
return 0;
}
static int snd_cs5535audio_capture_close(struct snd_pcm_substream *substream)
{
struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
olpc_capture_close(cs5535au->ac97);
return 0;
}

494
sound/pci/emu10k1/emu10k1_main.c
View File

File diff suppressed because it is too large Load Diff

46
sound/pci/emu10k1/emumixer.c
View File

@ -1639,6 +1639,45 @@ static struct snd_kcontrol_new snd_audigy_shared_spdif __devinitdata =
.put = snd_emu10k1_shared_spdif_put
};
/* workaround for too low volume on Audigy due to 16bit/24bit conversion */
#define snd_audigy_capture_boost_info snd_ctl_boolean_mono_info
static int snd_audigy_capture_boost_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
/* FIXME: better to use a cached version */
val = snd_ac97_read(emu->ac97, AC97_REC_GAIN);
ucontrol->value.integer.value[0] = !!val;
return 0;
}
static int snd_audigy_capture_boost_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
if (ucontrol->value.integer.value[0])
val = 0x0f0f;
else
val = 0;
return snd_ac97_update(emu->ac97, AC97_REC_GAIN, val);
}
static struct snd_kcontrol_new snd_audigy_capture_boost __devinitdata =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Analog Capture Boost",
.info = snd_audigy_capture_boost_info,
.get = snd_audigy_capture_boost_get,
.put = snd_audigy_capture_boost_put
};
/*
*/
static void snd_emu10k1_mixer_free_ac97(struct snd_ac97 *ac97)
@ -2087,5 +2126,12 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
}
}
if (emu->card_capabilities->ac97_chip && emu->audigy) {
err = snd_ctl_add(card, snd_ctl_new1(&snd_audigy_capture_boost,
emu));
if (err < 0)
return err;
}
return 0;
}

2
sound/pci/es1968.c
View File

@ -1953,7 +1953,7 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
outw(inw(chip->io_port + 4) & 1, chip->io_port + 4);
if (event & ESM_HWVOL_IRQ)
tasklet_hi_schedule(&chip->hwvol_tq); /* we'll do this later */
tasklet_schedule(&chip->hwvol_tq); /* we'll do this later */
/* else ack 'em all, i imagine */
outb(0xFF, chip->io_port + 0x1A);

188
sound/pci/hda/Kconfig Normal file
View File

@ -0,0 +1,188 @@
menuconfig SND_HDA_INTEL
tristate "Intel HD Audio"
select SND_PCM
select SND_VMASTER
select SND_JACK if INPUT=y || INPUT=SND
help
Say Y here to include support for Intel "High Definition
Audio" (Azalia) and its compatible devices.
This option enables the HD-audio controller. Don't forget
to choose the appropriate codec options below.
To compile this driver as a module, choose M here: the module
will be called snd-hda-intel.
if SND_HDA_INTEL
config SND_HDA_HWDEP
bool "Build hwdep interface for HD-audio driver"
select SND_HWDEP
help
Say Y here to build a hwdep interface for HD-audio driver.
This interface can be used for out-of-band communication
with codecs for debugging purposes.
config SND_HDA_RECONFIG
bool "Allow dynamic codec reconfiguration (EXPERIMENTAL)"
depends on SND_HDA_HWDEP && EXPERIMENTAL
help
Say Y here to enable the HD-audio codec re-configuration feature.
This adds the sysfs interfaces to allow user to clear the whole
codec configuration, change the codec setup, add extra verbs,
and re-configure the codec dynamically.
config SND_HDA_INPUT_BEEP
bool "Support digital beep via input layer"
depends on INPUT=y || INPUT=SND_HDA_INTEL
help
Say Y here to build a digital beep interface for HD-audio
driver. This interface is used to generate digital beeps.
config SND_HDA_CODEC_REALTEK
bool "Build Realtek HD-audio codec support"
default y
help
Say Y here to include Realtek HD-audio codec support in
snd-hda-intel driver, such as ALC880.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-realtek.
This module is automatically loaded at probing.
config SND_HDA_CODEC_ANALOG
bool "Build Analog Device HD-audio codec support"
default y
help
Say Y here to include Analog Device HD-audio codec support in
snd-hda-intel driver, such as AD1986A.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-analog.
This module is automatically loaded at probing.
config SND_HDA_CODEC_SIGMATEL
bool "Build IDT/Sigmatel HD-audio codec support"
default y
help
Say Y here to include IDT (Sigmatel) HD-audio codec support in
snd-hda-intel driver, such as STAC9200.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-idt.
This module is automatically loaded at probing.
config SND_HDA_CODEC_VIA
bool "Build VIA HD-audio codec support"
default y
help
Say Y here to include VIA HD-audio codec support in
snd-hda-intel driver, such as VT1708.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-via.
This module is automatically loaded at probing.
config SND_HDA_CODEC_ATIHDMI
bool "Build ATI HDMI HD-audio codec support"
default y
help
Say Y here to include ATI HDMI HD-audio codec support in
snd-hda-intel driver, such as ATI RS600 HDMI.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-atihdmi.
This module is automatically loaded at probing.
config SND_HDA_CODEC_NVHDMI
bool "Build NVIDIA HDMI HD-audio codec support"
default y
help
Say Y here to include NVIDIA HDMI HD-audio codec support in
snd-hda-intel driver, such as NVIDIA MCP78 HDMI.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-nvhdmi.
This module is automatically loaded at probing.
config SND_HDA_CODEC_INTELHDMI
bool "Build INTEL HDMI HD-audio codec support"
default y
help
Say Y here to include INTEL HDMI HD-audio codec support in
snd-hda-intel driver, such as Eaglelake integrated HDMI.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-intelhdmi.
This module is automatically loaded at probing.
config SND_HDA_ELD
def_bool y
depends on SND_HDA_CODEC_INTELHDMI
config SND_HDA_CODEC_CONEXANT
bool "Build Conexant HD-audio codec support"
default y
help
Say Y here to include Conexant HD-audio codec support in
snd-hda-intel driver, such as CX20549.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-conexant.
This module is automatically loaded at probing.
config SND_HDA_CODEC_CMEDIA
bool "Build C-Media HD-audio codec support"
default y
help
Say Y here to include C-Media HD-audio codec support in
snd-hda-intel driver, such as CMI9880.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-cmedia.
This module is automatically loaded at probing.
config SND_HDA_CODEC_SI3054
bool "Build Silicon Labs 3054 HD-modem codec support"
default y
help
Say Y here to include Silicon Labs 3054 HD-modem codec
(and compatibles) support in snd-hda-intel driver.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-si3054.
This module is automatically loaded at probing.
config SND_HDA_GENERIC
bool "Enable generic HD-audio codec parser"
default y
help
Say Y here to enable the generic HD-audio codec parser
in snd-hda-intel driver.
config SND_HDA_POWER_SAVE
bool "Aggressive power-saving on HD-audio"
help
Say Y here to enable more aggressive power-saving mode on
HD-audio driver. The power-saving timeout can be configured
via power_save option or over sysfs on-the-fly.
config SND_HDA_POWER_SAVE_DEFAULT
int "Default time-out for HD-audio power-save mode"
depends on SND_HDA_POWER_SAVE
default 0
help
The default time-out value in seconds for HD-audio automatic
power-save mode. 0 means to disable the power-save mode.
endif

75
sound/pci/hda/Makefile
View File

@ -1,20 +1,59 @@
snd-hda-intel-y := hda_intel.o
# since snd-hda-intel is the only driver using hda-codec,
# merge it into a single module although it was originally
# designed to be individual modules
snd-hda-intel-y += hda_codec.o
snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o
snd-hda-intel-$(CONFIG_SND_HDA_CODEC_NVHDMI) += patch_nvhdmi.o
snd-hda-intel-objs := hda_intel.o
snd-hda-codec-y := hda_codec.o
snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
# snd-hda-codec-$(CONFIG_SND_HDA_ELD) += hda_eld.o
snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
snd-hda-codec-realtek-objs := patch_realtek.o
snd-hda-codec-cmedia-objs := patch_cmedia.o
snd-hda-codec-analog-objs := patch_analog.o
snd-hda-codec-idt-objs := patch_sigmatel.o
snd-hda-codec-si3054-objs := patch_si3054.o
snd-hda-codec-atihdmi-objs := patch_atihdmi.o
snd-hda-codec-conexant-objs := patch_conexant.o
snd-hda-codec-via-objs := patch_via.o
snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o
snd-hda-codec-intelhdmi-objs := patch_intelhdmi.o hda_eld.o
# common driver
obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o
# codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans)
ifdef CONFIG_SND_HDA_CODEC_REALTEK
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o
endif
ifdef CONFIG_SND_HDA_CODEC_CMEDIA
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cmedia.o
endif
ifdef CONFIG_SND_HDA_CODEC_ANALOG
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-analog.o
endif
ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-idt.o
endif
ifdef CONFIG_SND_HDA_CODEC_SI3054
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o
endif
ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o
endif
ifdef CONFIG_SND_HDA_CODEC_CONEXANT
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
endif
ifdef CONFIG_SND_HDA_CODEC_VIA
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o
endif
ifdef CONFIG_SND_HDA_CODEC_NVHDMI
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-nvhdmi.o
endif
ifdef CONFIG_SND_HDA_CODEC_INTELHDMI
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-intelhdmi.o
endif
# this must be the last entry after codec drivers;
# otherwise the codec patches won't be hooked before the PCI probe
# when built in kernel
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o

2
sound/pci/hda/hda_beep.c
View File

@ -128,6 +128,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device);
void snd_hda_detach_beep_device(struct hda_codec *codec)
{
@ -140,3 +141,4 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
kfree(beep);
}
}
EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);

781
sound/pci/hda/hda_codec.c
View File

File diff suppressed because it is too large Load Diff

107
sound/pci/hda/hda_codec.h
View File

@ -519,6 +519,36 @@ enum {
/* max. codec address */
#define HDA_MAX_CODEC_ADDRESS 0x0f
/*
* generic arrays
*/
struct snd_array {
unsigned int used;
unsigned int alloced;
unsigned int elem_size;
unsigned int alloc_align;
void *list;
};
void *snd_array_new(struct snd_array *array);
void snd_array_free(struct snd_array *array);
static inline void snd_array_init(struct snd_array *array, unsigned int size,
unsigned int align)
{
array->elem_size = size;
array->alloc_align = align;
}
static inline void *snd_array_elem(struct snd_array *array, unsigned int idx)
{
return array->list + idx * array->elem_size;
}
static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
{
return (unsigned long)(ptr - array->list) / array->elem_size;
}
/*
* Structures
*/
@ -536,15 +566,17 @@ typedef u16 hda_nid_t;
/* bus operators */
struct hda_bus_ops {
/* send a single command */
int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
unsigned int verb, unsigned int parm);
int (*command)(struct hda_bus *bus, unsigned int cmd);
/* get a response from the last command */
unsigned int (*get_response)(struct hda_codec *codec);
unsigned int (*get_response)(struct hda_bus *bus);
/* free the private data */
void (*private_free)(struct hda_bus *);
/* attach a PCM stream */
int (*attach_pcm)(struct hda_bus *bus, struct hda_codec *codec,
struct hda_pcm *pcm);
#ifdef CONFIG_SND_HDA_POWER_SAVE
/* notify power-up/down from codec to controller */
void (*pm_notify)(struct hda_codec *codec);
void (*pm_notify)(struct hda_bus *bus);
#endif
};
@ -553,6 +585,7 @@ struct hda_bus_template {
void *private_data;
struct pci_dev *pci;
const char *modelname;
int *power_save;
struct hda_bus_ops ops;
};
@ -569,6 +602,7 @@ struct hda_bus {
void *private_data;
struct pci_dev *pci;
const char *modelname;
int *power_save;
struct hda_bus_ops ops;
/* codec linked list */
@ -581,10 +615,12 @@ struct hda_bus {
/* unsolicited event queue */
struct hda_bus_unsolicited *unsol;
struct snd_info_entry *proc;
/* assigned PCMs */
DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES);
/* misc op flags */
unsigned int needs_damn_long_delay :1;
unsigned int shutdown :1; /* being unloaded */
};
/*
@ -604,6 +640,16 @@ struct hda_codec_preset {
int (*patch)(struct hda_codec *codec);
};
struct hda_codec_preset_list {
const struct hda_codec_preset *preset;
struct module *owner;
struct list_head list;
};
/* initial hook */
int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset);
int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset);
/* ops set by the preset patch */
struct hda_codec_ops {
int (*build_controls)(struct hda_codec *codec);
@ -635,10 +681,7 @@ struct hda_amp_info {
struct hda_cache_rec {
u16 hash[64]; /* hash table for index */
unsigned int num_entries; /* number of assigned entries */
unsigned int size; /* allocated size */
unsigned int record_size; /* record size (including header) */
void *buffer; /* hash table entries */
struct snd_array buf; /* record entries */
};
/* PCM callbacks */
@ -680,7 +723,8 @@ struct hda_pcm {
char *name;
struct hda_pcm_stream stream[2];
unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */
int device; /* assigned device number */
int device; /* device number to assign */
struct snd_pcm *pcm; /* assigned PCM instance */
};
/* codec information */
@ -699,6 +743,9 @@ struct hda_codec {
/* detected preset */
const struct hda_codec_preset *preset;
struct module *owner;
const char *name; /* codec name */
const char *modelname; /* model name for preset */
/* set by patch */
struct hda_codec_ops patch_ops;
@ -718,6 +765,8 @@ struct hda_codec {
hda_nid_t start_nid;
u32 *wcaps;
struct snd_array mixers; /* list of assigned mixer elements */
struct hda_cache_rec amp_cache; /* cache for amp access */
struct hda_cache_rec cmd_cache; /* cache for other commands */
@ -727,7 +776,11 @@ struct hda_codec {
unsigned int spdif_in_enable; /* SPDIF input enable? */
hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
#ifdef CONFIG_SND_HDA_HWDEP
struct snd_hwdep *hwdep; /* assigned hwdep device */
struct snd_array init_verbs; /* additional init verbs */
struct snd_array hints; /* additional hints */
#endif
/* misc flags */
unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each
@ -740,6 +793,10 @@ struct hda_codec {
int power_count; /* current (global) power refcount */
struct delayed_work power_work; /* delayed task for powerdown */
#endif
/* codec-specific additional proc output */
void (*proc_widget_hook)(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid);
};
/* direction */
@ -754,7 +811,7 @@ enum {
int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
struct hda_bus **busp);
int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
struct hda_codec **codecp);
int do_init, struct hda_codec **codecp);
/*
* low level functions
@ -799,11 +856,13 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec);
* Mixer
*/
int snd_hda_build_controls(struct hda_bus *bus);
int snd_hda_codec_build_controls(struct hda_codec *codec);
/*
* PCM
*/
int snd_hda_build_pcms(struct hda_bus *bus);
int snd_hda_codec_build_pcms(struct hda_codec *codec);
void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
u32 stream_tag,
int channel_id, int format);
@ -812,8 +871,6 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
unsigned int channels,
unsigned int format,
unsigned int maxbps);
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
unsigned int format);
@ -830,6 +887,13 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state);
int snd_hda_resume(struct hda_bus *bus);
#endif
/*
* get widget information
*/
const char *snd_hda_get_jack_connectivity(u32 cfg);
const char *snd_hda_get_jack_type(u32 cfg);
const char *snd_hda_get_jack_location(u32 cfg);
/*
* power saving
*/
@ -837,12 +901,25 @@ int snd_hda_resume(struct hda_bus *bus);
void snd_hda_power_up(struct hda_codec *codec);
void snd_hda_power_down(struct hda_codec *codec);
#define snd_hda_codec_needs_resume(codec) codec->power_count
int snd_hda_codecs_inuse(struct hda_bus *bus);
#else
static inline void snd_hda_power_up(struct hda_codec *codec) {}
static inline void snd_hda_power_down(struct hda_codec *codec) {}
#define snd_hda_codec_needs_resume(codec) 1
#define snd_hda_codecs_inuse(bus) 1
#endif
/*
* Codec modularization
*/
/* Export symbols only for communication with codec drivers;
* When built in kernel, all HD-audio drivers are supposed to be statically
* linked to the kernel. Thus, the symbols don't have to (or shouldn't) be
* exported unless it's built as a module.
*/
#ifdef MODULE
#define EXPORT_SYMBOL_HDA(sym) EXPORT_SYMBOL_GPL(sym)
#else
#define EXPORT_SYMBOL_HDA(sym)
#endif
#endif /* __SOUND_HDA_CODEC_H */

590
sound/pci/hda/hda_eld.c Normal file
View File

@ -0,0 +1,590 @@
/*
* Generic routines and proc interface for ELD(EDID Like Data) information
*
* Copyright(c) 2008 Intel Corporation.
*
* Authors:
* Wu Fengguang <wfg@linux.intel.com>
*
* This driver is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This driver is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <linux/init.h>
#include <sound/core.h>
#include <asm/unaligned.h>
#include "hda_codec.h"
#include "hda_local.h"
enum eld_versions {
ELD_VER_CEA_861D = 2,
ELD_VER_PARTIAL = 31,
};
enum cea_edid_versions {
CEA_EDID_VER_NONE = 0,
CEA_EDID_VER_CEA861 = 1,
CEA_EDID_VER_CEA861A = 2,
CEA_EDID_VER_CEA861BCD = 3,
CEA_EDID_VER_RESERVED = 4,
};
static char *cea_speaker_allocation_names[] = {
/* 0 */ "FL/FR",
/* 1 */ "LFE",
/* 2 */ "FC",
/* 3 */ "RL/RR",
/* 4 */ "RC",
/* 5 */ "FLC/FRC",
/* 6 */ "RLC/RRC",
/* 7 */ "FLW/FRW",
/* 8 */ "FLH/FRH",
/* 9 */ "TC",
/* 10 */ "FCH",
};
static char *eld_connection_type_names[4] = {
"HDMI",
"DisplayPort",
"2-reserved",
"3-reserved"
};
enum cea_audio_coding_types {
AUDIO_CODING_TYPE_REF_STREAM_HEADER = 0,
AUDIO_CODING_TYPE_LPCM = 1,
AUDIO_CODING_TYPE_AC3 = 2,
AUDIO_CODING_TYPE_MPEG1 = 3,
AUDIO_CODING_TYPE_MP3 = 4,
AUDIO_CODING_TYPE_MPEG2 = 5,
AUDIO_CODING_TYPE_AACLC = 6,
AUDIO_CODING_TYPE_DTS = 7,
AUDIO_CODING_TYPE_ATRAC = 8,
AUDIO_CODING_TYPE_SACD = 9,
AUDIO_CODING_TYPE_EAC3 = 10,
AUDIO_CODING_TYPE_DTS_HD = 11,
AUDIO_CODING_TYPE_MLP = 12,
AUDIO_CODING_TYPE_DST = 13,
AUDIO_CODING_TYPE_WMAPRO = 14,
AUDIO_CODING_TYPE_REF_CXT = 15,
/* also include valid xtypes below */
AUDIO_CODING_TYPE_HE_AAC = 15,
AUDIO_CODING_TYPE_HE_AAC2 = 16,
AUDIO_CODING_TYPE_MPEG_SURROUND = 17,
};
enum cea_audio_coding_xtypes {
AUDIO_CODING_XTYPE_HE_REF_CT = 0,
AUDIO_CODING_XTYPE_HE_AAC = 1,
AUDIO_CODING_XTYPE_HE_AAC2 = 2,
AUDIO_CODING_XTYPE_MPEG_SURROUND = 3,
AUDIO_CODING_XTYPE_FIRST_RESERVED = 4,
};
static char *cea_audio_coding_type_names[] = {
/* 0 */ "undefined",
/* 1 */ "LPCM",
/* 2 */ "AC-3",
/* 3 */ "MPEG1",
/* 4 */ "MP3",
/* 5 */ "MPEG2",
/* 6 */ "AAC-LC",
/* 7 */ "DTS",
/* 8 */ "ATRAC",
/* 9 */ "DSD (One Bit Audio)",
/* 10 */ "E-AC-3/DD+ (Dolby Digital Plus)",
/* 11 */ "DTS-HD",
/* 12 */ "MLP (Dolby TrueHD)",
/* 13 */ "DST",
/* 14 */ "WMAPro",
/* 15 */ "HE-AAC",
/* 16 */ "HE-AACv2",
/* 17 */ "MPEG Surround",
};
/*
* The following two lists are shared between
* - HDMI audio InfoFrame (source to sink)
* - CEA E-EDID Extension (sink to source)
*/
/*
* SS1:SS0 index => sample size
*/
static int cea_sample_sizes[4] = {
0, /* 0: Refer to Stream Header */
AC_SUPPCM_BITS_16, /* 1: 16 bits */
AC_SUPPCM_BITS_20, /* 2: 20 bits */
AC_SUPPCM_BITS_24, /* 3: 24 bits */
};
/*
* SF2:SF1:SF0 index => sampling frequency
*/
static int cea_sampling_frequencies[8] = {
0, /* 0: Refer to Stream Header */
SNDRV_PCM_RATE_32000, /* 1: 32000Hz */
SNDRV_PCM_RATE_44100, /* 2: 44100Hz */
SNDRV_PCM_RATE_48000, /* 3: 48000Hz */
SNDRV_PCM_RATE_88200, /* 4: 88200Hz */
SNDRV_PCM_RATE_96000, /* 5: 96000Hz */
SNDRV_PCM_RATE_176400, /* 6: 176400Hz */
SNDRV_PCM_RATE_192000, /* 7: 192000Hz */
};
static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
int byte_index)
{
unsigned int val;
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_HDMI_ELDD, byte_index);
#ifdef BE_PARANOID
printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
#endif
if ((val & AC_ELDD_ELD_VALID) == 0) {
snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
byte_index);
val = 0;
}
return val & AC_ELDD_ELD_DATA;
}
#define GRAB_BITS(buf, byte, lowbit, bits) \
({ \
BUILD_BUG_ON(lowbit > 7); \
BUILD_BUG_ON(bits > 8); \
BUILD_BUG_ON(bits <= 0); \
\
(buf[byte] >> (lowbit)) & ((1 << (bits)) - 1); \
})
static void hdmi_update_short_audio_desc(struct cea_sad *a,
const unsigned char *buf)
{
int i;
int val;
val = GRAB_BITS(buf, 1, 0, 7);
a->rates = 0;
for (i = 0; i < 7; i++)
if (val & (1 << i))
a->rates |= cea_sampling_frequencies[i + 1];
a->channels = GRAB_BITS(buf, 0, 0, 3);
a->channels++;
a->format = GRAB_BITS(buf, 0, 3, 4);
switch (a->format) {
case AUDIO_CODING_TYPE_REF_STREAM_HEADER:
snd_printd(KERN_INFO
"HDMI: audio coding type 0 not expected\n");
break;
case AUDIO_CODING_TYPE_LPCM:
val = GRAB_BITS(buf, 2, 0, 3);
a->sample_bits = 0;
for (i = 0; i < 3; i++)
if (val & (1 << i))
a->sample_bits |= cea_sample_sizes[i + 1];
break;
case AUDIO_CODING_TYPE_AC3:
case AUDIO_CODING_TYPE_MPEG1:
case AUDIO_CODING_TYPE_MP3:
case AUDIO_CODING_TYPE_MPEG2:
case AUDIO_CODING_TYPE_AACLC:
case AUDIO_CODING_TYPE_DTS:
case AUDIO_CODING_TYPE_ATRAC:
a->max_bitrate = GRAB_BITS(buf, 2, 0, 8);
a->max_bitrate *= 8000;
break;
case AUDIO_CODING_TYPE_SACD:
break;
case AUDIO_CODING_TYPE_EAC3:
break;
case AUDIO_CODING_TYPE_DTS_HD:
break;
case AUDIO_CODING_TYPE_MLP:
break;
case AUDIO_CODING_TYPE_DST:
break;
case AUDIO_CODING_TYPE_WMAPRO:
a->profile = GRAB_BITS(buf, 2, 0, 3);
break;
case AUDIO_CODING_TYPE_REF_CXT:
a->format = GRAB_BITS(buf, 2, 3, 5);
if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT ||
a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) {
snd_printd(KERN_INFO
"HDMI: audio coding xtype %d not expected\n",
a->format);
a->format = 0;
} else
a->format += AUDIO_CODING_TYPE_HE_AAC -
AUDIO_CODING_XTYPE_HE_AAC;
break;
}
}
/*
* Be careful, ELD buf could be totally rubbish!
*/
static int hdmi_update_eld(struct hdmi_eld *e,
const unsigned char *buf, int size)
{
int mnl;
int i;
e->eld_ver = GRAB_BITS(buf, 0, 3, 5);
if (e->eld_ver != ELD_VER_CEA_861D &&
e->eld_ver != ELD_VER_PARTIAL) {
snd_printd(KERN_INFO "HDMI: Unknown ELD version %d\n",
e->eld_ver);
goto out_fail;
}
e->eld_size = size;
e->baseline_len = GRAB_BITS(buf, 2, 0, 8);
mnl = GRAB_BITS(buf, 4, 0, 5);
e->cea_edid_ver = GRAB_BITS(buf, 4, 5, 3);
e->support_hdcp = GRAB_BITS(buf, 5, 0, 1);
e->support_ai = GRAB_BITS(buf, 5, 1, 1);
e->conn_type = GRAB_BITS(buf, 5, 2, 2);
e->sad_count = GRAB_BITS(buf, 5, 4, 4);
e->aud_synch_delay = GRAB_BITS(buf, 6, 0, 8) * 2;
e->spk_alloc = GRAB_BITS(buf, 7, 0, 7);
e->port_id = get_unaligned_le64(buf + 8);
/* not specified, but the spec's tendency is little endian */
e->manufacture_id = get_unaligned_le16(buf + 16);
e->product_id = get_unaligned_le16(buf + 18);
if (mnl > ELD_MAX_MNL) {
snd_printd(KERN_INFO "HDMI: MNL is reserved value %d\n", mnl);
goto out_fail;
} else if (ELD_FIXED_BYTES + mnl > size) {
snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl);
goto out_fail;
} else
strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl);
for (i = 0; i < e->sad_count; i++) {
if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {
snd_printd(KERN_INFO "HDMI: out of range SAD %d\n", i);
goto out_fail;
}
hdmi_update_short_audio_desc(e->sad + i,
buf + ELD_FIXED_BYTES + mnl + 3 * i);
}
return 0;
out_fail:
e->eld_ver = 0;
return -EINVAL;
}
static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid)
{
return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0);
}
static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid)
{
int eldv;
int present;
present = hdmi_present_sense(codec, nid);
eldv = (present & AC_PINSENSE_ELDV);
present = (present & AC_PINSENSE_PRESENCE);
#ifdef CONFIG_SND_DEBUG_VERBOSE
printk(KERN_INFO "HDMI: sink_present = %d, eld_valid = %d\n",
!!present, !!eldv);
#endif
return eldv && present;
}
int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid)
{
return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE,
AC_DIPSIZE_ELD_BUF);
}
int snd_hdmi_get_eld(struct hdmi_eld *eld,
struct hda_codec *codec, hda_nid_t nid)
{
int i;
int ret;
int size;
unsigned char *buf;
if (!hdmi_eld_valid(codec, nid))
return -ENOENT;
size = snd_hdmi_get_eld_size(codec, nid);
if (size == 0) {
/* wfg: workaround for ASUS P5E-VM HDMI board */
snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n");
size = 128;
}
if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) {
snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size);
return -ERANGE;
}
buf = kmalloc(size, GFP_KERNEL);
if (!buf)
return -ENOMEM;
for (i = 0; i < size; i++)
buf[i] = hdmi_get_eld_byte(codec, nid, i);
ret = hdmi_update_eld(eld, buf, size);
kfree(buf);
return ret;
}
static void hdmi_show_short_audio_desc(struct cea_sad *a)
{
char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
char buf2[8 + SND_PRINT_BITS_ADVISED_BUFSIZE] = ", bits =";
if (!a->format)
return;
snd_print_pcm_rates(a->rates, buf, sizeof(buf));
if (a->format == AUDIO_CODING_TYPE_LPCM)
snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
else if (a->max_bitrate)
snprintf(buf2, sizeof(buf2),
", max bitrate = %d", a->max_bitrate);
else
buf2[0] = '\0';
printk(KERN_INFO "HDMI: supports coding type %s:"
" channels = %d, rates =%s%s\n",
cea_audio_coding_type_names[a->format],
a->channels,
buf,
buf2);
}
void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen)
{
int i, j;
for (i = 0, j = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) {
if (spk_alloc & (1 << i))
j += snprintf(buf + j, buflen - j, " %s",
cea_speaker_allocation_names[i]);
}
buf[j] = '\0'; /* necessary when j == 0 */
}
void snd_hdmi_show_eld(struct hdmi_eld *e)
{
int i;
printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n",
e->monitor_name,
eld_connection_type_names[e->conn_type]);
if (e->spk_alloc) {
char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
printk(KERN_INFO "HDMI: available speakers:%s\n", buf);
}
for (i = 0; i < e->sad_count; i++)
hdmi_show_short_audio_desc(e->sad + i);
}
#ifdef CONFIG_PROC_FS
static void hdmi_print_sad_info(int i, struct cea_sad *a,
struct snd_info_buffer *buffer)
{
char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
snd_iprintf(buffer, "sad%d_coding_type\t[0x%x] %s\n",
i, a->format, cea_audio_coding_type_names[a->format]);
snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels);
snd_print_pcm_rates(a->rates, buf, sizeof(buf));
snd_iprintf(buffer, "sad%d_rates\t\t[0x%x]%s\n", i, a->rates, buf);
if (a->format == AUDIO_CODING_TYPE_LPCM) {
snd_print_pcm_bits(a->sample_bits, buf, sizeof(buf));
snd_iprintf(buffer, "sad%d_bits\t\t[0x%x]%s\n",
i, a->sample_bits, buf);
}
if (a->max_bitrate)
snd_iprintf(buffer, "sad%d_max_bitrate\t%d\n",
i, a->max_bitrate);
if (a->profile)
snd_iprintf(buffer, "sad%d_profile\t\t%d\n", i, a->profile);
}
static void hdmi_print_eld_info(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct hdmi_eld *e = entry->private_data;
char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
int i;
static char *eld_versoin_names[32] = {
"reserved",
"reserved",
"CEA-861D or below",
[3 ... 30] = "reserved",
[31] = "partial"
};
static char *cea_edid_version_names[8] = {
"no CEA EDID Timing Extension block present",
"CEA-861",
"CEA-861-A",
"CEA-861-B, C or D",
[4 ... 7] = "reserved"
};
snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name);
snd_iprintf(buffer, "connection_type\t\t%s\n",
eld_connection_type_names[e->conn_type]);
snd_iprintf(buffer, "eld_version\t\t[0x%x] %s\n", e->eld_ver,
eld_versoin_names[e->eld_ver]);
snd_iprintf(buffer, "edid_version\t\t[0x%x] %s\n", e->cea_edid_ver,
cea_edid_version_names[e->cea_edid_ver]);
snd_iprintf(buffer, "manufacture_id\t\t0x%x\n", e->manufacture_id);
snd_iprintf(buffer, "product_id\t\t0x%x\n", e->product_id);
snd_iprintf(buffer, "port_id\t\t\t0x%llx\n", (long long)e->port_id);
snd_iprintf(buffer, "support_hdcp\t\t%d\n", e->support_hdcp);
snd_iprintf(buffer, "support_ai\t\t%d\n", e->support_ai);
snd_iprintf(buffer, "audio_sync_delay\t%d\n", e->aud_synch_delay);
snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
snd_iprintf(buffer, "speakers\t\t[0x%x]%s\n", e->spk_alloc, buf);
snd_iprintf(buffer, "sad_count\t\t%d\n", e->sad_count);
for (i = 0; i < e->sad_count; i++)
hdmi_print_sad_info(i, e->sad + i, buffer);
}
static void hdmi_write_eld_info(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct hdmi_eld *e = entry->private_data;
char line[64];
char name[64];
char *sname;
long long val;
int n;
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%s %llx", name, &val) != 2)
continue;
/*
* We don't allow modification to these fields:
* monitor_name manufacture_id product_id
* eld_version edid_version
*/
if (!strcmp(name, "connection_type"))
e->conn_type = val;
else if (!strcmp(name, "port_id"))
e->port_id = val;
else if (!strcmp(name, "support_hdcp"))
e->support_hdcp = val;
else if (!strcmp(name, "support_ai"))
e->support_ai = val;
else if (!strcmp(name, "audio_sync_delay"))
e->aud_synch_delay = val;
else if (!strcmp(name, "speakers"))
e->spk_alloc = val;
else if (!strcmp(name, "sad_count"))
e->sad_count = val;
else if (!strncmp(name, "sad", 3)) {
sname = name + 4;
n = name[3] - '0';
if (name[4] >= '0' && name[4] <= '9') {
sname++;
n = 10 * n + name[4] - '0';
}
if (n < 0 || n > 31) /* double the CEA limit */
continue;
if (!strcmp(sname, "_coding_type"))
e->sad[n].format = val;
else if (!strcmp(sname, "_channels"))
e->sad[n].channels = val;
else if (!strcmp(sname, "_rates"))
e->sad[n].rates = val;
else if (!strcmp(sname, "_bits"))
e->sad[n].sample_bits = val;
else if (!strcmp(sname, "_max_bitrate"))
e->sad[n].max_bitrate = val;
else if (!strcmp(sname, "_profile"))
e->sad[n].profile = val;
if (n >= e->sad_count)
e->sad_count = n + 1;
}
}
}
int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld)
{
char name[32];
struct snd_info_entry *entry;
int err;
snprintf(name, sizeof(name), "eld#%d", codec->addr);
err = snd_card_proc_new(codec->bus->card, name, &entry);
if (err < 0)
return err;
snd_info_set_text_ops(entry, eld, hdmi_print_eld_info);
entry->c.text.write = hdmi_write_eld_info;
entry->mode |= S_IWUSR;
eld->proc_entry = entry;
return 0;
}
void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
{
if (!codec->bus->shutdown && eld->proc_entry) {
snd_device_free(codec->bus->card, eld->proc_entry);
eld->proc_entry = NULL;
}
}
#endif /* CONFIG_PROC_FS */

21
sound/pci/hda/hda_generic.c
View File

@ -723,7 +723,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
if (is_loopback)
add_input_loopback(codec, node->nid, HDA_INPUT, index);
snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
@ -732,7 +733,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
if (is_loopback)
add_input_loopback(codec, node->nid, HDA_OUTPUT, 0);
snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
}
@ -745,14 +747,16 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
(node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT);
snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
(node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT);
snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
}
@ -849,8 +853,8 @@ static int build_input_controls(struct hda_codec *codec)
}
/* create input MUX if multiple sources are available */
if ((err = snd_ctl_add(codec->bus->card,
snd_ctl_new1(&cap_sel, codec))) < 0)
err = snd_hda_ctl_add(codec, snd_ctl_new1(&cap_sel, codec));
if (err < 0)
return err;
/* no volume control? */
@ -867,8 +871,8 @@ static int build_input_controls(struct hda_codec *codec)
HDA_CODEC_VOLUME(name, adc_node->nid,
spec->input_mux.items[i].index,
HDA_INPUT);
if ((err = snd_ctl_add(codec->bus->card,
snd_ctl_new1(&knew, codec))) < 0)
err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
}
@ -1097,3 +1101,4 @@ int snd_hda_parse_generic_codec(struct hda_codec *codec)
snd_hda_generic_free(codec);
return err;
}
EXPORT_SYMBOL(snd_hda_parse_generic_codec);

234
sound/pci/hda/hda_hwdep.c
View File

@ -23,10 +23,12 @@
#include <linux/pci.h>
#include <linux/compat.h>
#include <linux/mutex.h>
#include <linux/ctype.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#include <sound/hda_hwdep.h>
#include <sound/minors.h>
/*
* write/read an out-of-bound verb
@ -95,7 +97,26 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
return 0;
}
int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
static void clear_hwdep_elements(struct hda_codec *codec)
{
char **head;
int i;
/* clear init verbs */
snd_array_free(&codec->init_verbs);
/* clear hints */
head = codec->hints.list;
for (i = 0; i < codec->hints.used; i++, head++)
kfree(*head);
snd_array_free(&codec->hints);
}
static void hwdep_free(struct snd_hwdep *hwdep)
{
clear_hwdep_elements(hwdep->private_data);
}
int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
{
char hwname[16];
struct snd_hwdep *hwdep;
@ -109,6 +130,7 @@ int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
sprintf(hwdep->name, "HDA Codec %d", codec->addr);
hwdep->iface = SNDRV_HWDEP_IFACE_HDA;
hwdep->private_data = codec;
hwdep->private_free = hwdep_free;
hwdep->exclusive = 1;
hwdep->ops.open = hda_hwdep_open;
@ -117,5 +139,215 @@ int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat;
#endif
snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32);
snd_array_init(&codec->hints, sizeof(char *), 32);
return 0;
}
#ifdef CONFIG_SND_HDA_RECONFIG
/*
* sysfs interface
*/
static int clear_codec(struct hda_codec *codec)
{
snd_hda_codec_reset(codec);
clear_hwdep_elements(codec);
return 0;
}
static int reconfig_codec(struct hda_codec *codec)
{
int err;
snd_printk(KERN_INFO "hda-codec: reconfiguring\n");
snd_hda_codec_reset(codec);
err = snd_hda_codec_configure(codec);
if (err < 0)
return err;
/* rebuild PCMs */
err = snd_hda_codec_build_pcms(codec);
if (err < 0)
return err;
/* rebuild mixers */
err = snd_hda_codec_build_controls(codec);
if (err < 0)
return err;
return 0;
}
/*
* allocate a string at most len chars, and remove the trailing EOL
*/
static char *kstrndup_noeol(const char *src, size_t len)
{
char *s = kstrndup(src, len, GFP_KERNEL);
char *p;
if (!s)
return NULL;
p = strchr(s, '\n');
if (p)
*p = 0;
return s;
}
#define CODEC_INFO_SHOW(type) \
static ssize_t type##_show(struct device *dev, \
struct device_attribute *attr, \
char *buf) \
{ \
struct snd_hwdep *hwdep = dev_get_drvdata(dev); \
struct hda_codec *codec = hwdep->private_data; \
return sprintf(buf, "0x%x\n", codec->type); \
}
#define CODEC_INFO_STR_SHOW(type) \
static ssize_t type##_show(struct device *dev, \
struct device_attribute *attr, \
char *buf) \
{ \
struct snd_hwdep *hwdep = dev_get_drvdata(dev); \
struct hda_codec *codec = hwdep->private_data; \
return sprintf(buf, "%s\n", \
codec->type ? codec->type : ""); \
}
CODEC_INFO_SHOW(vendor_id);
CODEC_INFO_SHOW(subsystem_id);
CODEC_INFO_SHOW(revision_id);
CODEC_INFO_SHOW(afg);
CODEC_INFO_SHOW(mfg);
CODEC_INFO_STR_SHOW(name);
CODEC_INFO_STR_SHOW(modelname);
#define CODEC_INFO_STORE(type) \
static ssize_t type##_store(struct device *dev, \
struct device_attribute *attr, \
const char *buf, size_t count) \
{ \
struct snd_hwdep *hwdep = dev_get_drvdata(dev); \
struct hda_codec *codec = hwdep->private_data; \
char *after; \
codec->type = simple_strtoul(buf, &after, 0); \
return count; \
}
#define CODEC_INFO_STR_STORE(type) \
static ssize_t type##_store(struct device *dev, \
struct device_attribute *attr, \
const char *buf, size_t count) \
{ \
struct snd_hwdep *hwdep = dev_get_drvdata(dev); \
struct hda_codec *codec = hwdep->private_data; \
char *s = kstrndup_noeol(buf, 64); \
if (!s) \
return -ENOMEM; \
kfree(codec->type); \
codec->type = s; \
return count; \
}
CODEC_INFO_STORE(vendor_id);
CODEC_INFO_STORE(subsystem_id);
CODEC_INFO_STORE(revision_id);
CODEC_INFO_STR_STORE(name);
CODEC_INFO_STR_STORE(modelname);
#define CODEC_ACTION_STORE(type) \
static ssize_t type##_store(struct device *dev, \
struct device_attribute *attr, \
const char *buf, size_t count) \
{ \
struct snd_hwdep *hwdep = dev_get_drvdata(dev); \
struct hda_codec *codec = hwdep->private_data; \
int err = 0; \
if (*buf) \
err = type##_codec(codec); \
return err < 0 ? err : count; \
}
CODEC_ACTION_STORE(reconfig);
CODEC_ACTION_STORE(clear);
static ssize_t init_verbs_store(struct device *dev,
struct device_attribute *attr,
const char *buf, size_t count)
{
struct snd_hwdep *hwdep = dev_get_drvdata(dev);
struct hda_codec *codec = hwdep->private_data;
char *p;
struct hda_verb verb, *v;
verb.nid = simple_strtoul(buf, &p, 0);
verb.verb = simple_strtoul(p, &p, 0);
verb.param = simple_strtoul(p, &p, 0);
if (!verb.nid || !verb.verb || !verb.param)
return -EINVAL;
v = snd_array_new(&codec->init_verbs);
if (!v)
return -ENOMEM;
*v = verb;
return count;
}
static ssize_t hints_store(struct device *dev,
struct device_attribute *attr,
const char *buf, size_t count)
{
struct snd_hwdep *hwdep = dev_get_drvdata(dev);
struct hda_codec *codec = hwdep->private_data;
char *p;
char **hint;
if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n')
return count;
p = kstrndup_noeol(buf, 1024);
if (!p)
return -ENOMEM;
hint = snd_array_new(&codec->hints);
if (!hint) {
kfree(p);
return -ENOMEM;
}
*hint = p;
return count;
}
#define CODEC_ATTR_RW(type) \
__ATTR(type, 0644, type##_show, type##_store)
#define CODEC_ATTR_RO(type) \
__ATTR_RO(type)
#define CODEC_ATTR_WO(type) \
__ATTR(type, 0200, NULL, type##_store)
static struct device_attribute codec_attrs[] = {
CODEC_ATTR_RW(vendor_id),
CODEC_ATTR_RW(subsystem_id),
CODEC_ATTR_RW(revision_id),
CODEC_ATTR_RO(afg),
CODEC_ATTR_RO(mfg),
CODEC_ATTR_RW(name),
CODEC_ATTR_RW(modelname),
CODEC_ATTR_WO(init_verbs),
CODEC_ATTR_WO(hints),
CODEC_ATTR_WO(reconfig),
CODEC_ATTR_WO(clear),
};
/*
* create sysfs files on hwdep directory
*/
int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
{
struct snd_hwdep *hwdep = codec->hwdep;
int i;
for (i = 0; i < ARRAY_SIZE(codec_attrs); i++)
snd_add_device_sysfs_file(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card,
hwdep->device, &codec_attrs[i]);
return 0;
}
#endif /* CONFIG_SND_HDA_RECONFIG */

346
sound/pci/hda/hda_intel.c
View File

@ -58,6 +58,7 @@ static char *model[SNDRV_CARDS];
static int position_fix[SNDRV_CARDS];
static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
static int probe_only[SNDRV_CARDS];
static int single_cmd;
static int enable_msi;
@ -76,6 +77,8 @@ module_param_array(bdl_pos_adj, int, NULL, 0644);
MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
module_param_array(probe_mask, int, NULL, 0444);
MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1).");
module_param_array(probe_only, bool, NULL, 0444);
MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization.");
module_param(single_cmd, bool, 0444);
MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
"(for debugging only).");
@ -83,7 +86,10 @@ module_param(enable_msi, int, 0444);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
#ifdef CONFIG_SND_HDA_POWER_SAVE
/* power_save option is defined in hda_codec.c */
static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
module_param(power_save, int, 0644);
MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
"(in second, 0 = disable).");
/* reset the HD-audio controller in power save mode.
* this may give more power-saving, but will take longer time to
@ -292,6 +298,8 @@ enum {
/* Define VIA HD Audio Device ID*/
#define VIA_HDAC_DEVICE_ID 0x3288
/* HD Audio class code */
#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403
/*
*/
@ -392,6 +400,7 @@ struct azx {
unsigned int msi :1;
unsigned int irq_pending_warned :1;
unsigned int via_dmapos_patch :1; /* enable DMA-position fix for VIA */
unsigned int probing :1; /* codec probing phase */
/* for debugging */
unsigned int last_cmd; /* last issued command (to sync) */
@ -414,6 +423,7 @@ enum {
AZX_DRIVER_ULI,
AZX_DRIVER_NVIDIA,
AZX_DRIVER_TERA,
AZX_DRIVER_GENERIC,
AZX_NUM_DRIVERS, /* keep this as last entry */
};
@ -427,6 +437,7 @@ static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_ULI] = "HDA ULI M5461",
[AZX_DRIVER_NVIDIA] = "HDA NVidia",
[AZX_DRIVER_TERA] = "HDA Teradici",
[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
};
/*
@ -527,9 +538,9 @@ static void azx_free_cmd_io(struct azx *chip)
}
/* send a command */
static int azx_corb_send_cmd(struct hda_codec *codec, u32 val)
static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
{
struct azx *chip = codec->bus->private_data;
struct azx *chip = bus->private_data;
unsigned int wp;
/* add command to corb */
@ -577,9 +588,9 @@ static void azx_update_rirb(struct azx *chip)
}
/* receive a response */
static unsigned int azx_rirb_get_response(struct hda_codec *codec)
static unsigned int azx_rirb_get_response(struct hda_bus *bus)
{
struct azx *chip = codec->bus->private_data;
struct azx *chip = bus->private_data;
unsigned long timeout;
again:
@ -596,7 +607,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
}
if (time_after(jiffies, timeout))
break;
if (codec->bus->needs_damn_long_delay)
if (bus->needs_damn_long_delay)
msleep(2); /* temporary workaround */
else {
udelay(10);
@ -624,6 +635,14 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
goto again;
}
if (chip->probing) {
/* If this critical timeout happens during the codec probing
* phase, this is likely an access to a non-existing codec
* slot. Better to return an error and reset the system.
*/
return -1;
}
snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
"switching to single_cmd mode: last cmd=0x%08x\n",
chip->last_cmd);
@ -646,9 +665,9 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
*/
/* send a command */
static int azx_single_send_cmd(struct hda_codec *codec, u32 val)
static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
{
struct azx *chip = codec->bus->private_data;
struct azx *chip = bus->private_data;
int timeout = 50;
while (timeout--) {
@ -671,9 +690,9 @@ static int azx_single_send_cmd(struct hda_codec *codec, u32 val)
}
/* receive a response */
static unsigned int azx_single_get_response(struct hda_codec *codec)
static unsigned int azx_single_get_response(struct hda_bus *bus)
{
struct azx *chip = codec->bus->private_data;
struct azx *chip = bus->private_data;
int timeout = 50;
while (timeout--) {
@ -696,38 +715,29 @@ static unsigned int azx_single_get_response(struct hda_codec *codec)
*/
/* send a command */
static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid,
int direct, unsigned int verb,
unsigned int para)
static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
{
struct azx *chip = codec->bus->private_data;
u32 val;
struct azx *chip = bus->private_data;
val = (u32)(codec->addr & 0x0f) << 28;
val |= (u32)direct << 27;
val |= (u32)nid << 20;
val |= verb << 8;
val |= para;
chip->last_cmd = val;
if (chip->single_cmd)
return azx_single_send_cmd(codec, val);
return azx_single_send_cmd(bus, val);
else
return azx_corb_send_cmd(codec, val);
return azx_corb_send_cmd(bus, val);
}
/* get a response */
static unsigned int azx_get_response(struct hda_codec *codec)
static unsigned int azx_get_response(struct hda_bus *bus)
{
struct azx *chip = codec->bus->private_data;
struct azx *chip = bus->private_data;
if (chip->single_cmd)
return azx_single_get_response(codec);
return azx_single_get_response(bus);
else
return azx_rirb_get_response(codec);
return azx_rirb_get_response(bus);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static void azx_power_notify(struct hda_codec *codec);
static void azx_power_notify(struct hda_bus *bus);
#endif
/* reset codec link */
@ -1184,6 +1194,28 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
return 0;
}
/*
* Probe the given codec address
*/
static int probe_codec(struct azx *chip, int addr)
{
unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) |
(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
unsigned int res;
chip->probing = 1;
azx_send_cmd(chip->bus, cmd);
res = azx_get_response(chip->bus);
chip->probing = 0;
if (res == -1)
return -EIO;
snd_printdd("hda_intel: codec #%d probed OK\n", addr);
return 0;
}
static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
struct hda_pcm *cpcm);
static void azx_stop_chip(struct azx *chip);
/*
* Codec initialization
@ -1194,21 +1226,13 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
[AZX_DRIVER_TERA] = 1,
};
/* number of slots to probe as default
* this can be different from azx_max_codecs[] -- e.g. some boards
* report wrongly the non-existing 4th slot availability
*/
static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = {
[AZX_DRIVER_ICH] = 3,
[AZX_DRIVER_ATI] = 3,
};
static int __devinit azx_codec_create(struct azx *chip, const char *model,
unsigned int codec_probe_mask)
unsigned int codec_probe_mask,
int no_init)
{
struct hda_bus_template bus_temp;
int c, codecs, audio_codecs, err;
int def_slots, max_slots;
int c, codecs, err;
int max_slots;
memset(&bus_temp, 0, sizeof(bus_temp));
bus_temp.private_data = chip;
@ -1216,7 +1240,9 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
bus_temp.pci = chip->pci;
bus_temp.ops.command = azx_send_cmd;
bus_temp.ops.get_response = azx_get_response;
bus_temp.ops.attach_pcm = azx_attach_pcm_stream;
#ifdef CONFIG_SND_HDA_POWER_SAVE
bus_temp.power_save = &power_save;
bus_temp.ops.pm_notify = azx_power_notify;
#endif
@ -1227,33 +1253,43 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
if (chip->driver_type == AZX_DRIVER_NVIDIA)
chip->bus->needs_damn_long_delay = 1;
codecs = audio_codecs = 0;
codecs = 0;
max_slots = azx_max_codecs[chip->driver_type];
if (!max_slots)
max_slots = AZX_MAX_CODECS;
def_slots = azx_default_codecs[chip->driver_type];
if (!def_slots)
def_slots = max_slots;
for (c = 0; c < def_slots; c++) {
/* First try to probe all given codec slots */
for (c = 0; c < max_slots; c++) {
if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
if (probe_codec(chip, c) < 0) {
/* Some BIOSen give you wrong codec addresses
* that don't exist
*/
snd_printk(KERN_WARNING
"hda_intel: Codec #%d probe error; "
"disabling it...\n", c);
chip->codec_mask &= ~(1 << c);
/* More badly, accessing to a non-existing
* codec often screws up the controller chip,
* and distrubs the further communications.
* Thus if an error occurs during probing,
* better to reset the controller chip to
* get back to the sanity state.
*/
azx_stop_chip(chip);
azx_init_chip(chip);
}
}
}
/* Then create codec instances */
for (c = 0; c < max_slots; c++) {
if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
struct hda_codec *codec;
err = snd_hda_codec_new(chip->bus, c, &codec);
err = snd_hda_codec_new(chip->bus, c, !no_init, &codec);
if (err < 0)
continue;
codecs++;
if (codec->afg)
audio_codecs++;
}
}
if (!audio_codecs) {
/* probe additional slots if no codec is found */
for (; c < max_slots; c++) {
if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
err = snd_hda_codec_new(chip->bus, c, NULL);
if (err < 0)
continue;
codecs++;
}
}
}
if (!codecs) {
@ -1722,111 +1758,59 @@ static struct snd_pcm_ops azx_pcm_ops = {
static void azx_pcm_free(struct snd_pcm *pcm)
{
kfree(pcm->private_data);
struct azx_pcm *apcm = pcm->private_data;
if (apcm) {
apcm->chip->pcm[pcm->device] = NULL;
kfree(apcm);
}
}
static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
struct hda_pcm *cpcm)
static int
azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
struct hda_pcm *cpcm)
{
int err;
struct azx *chip = bus->private_data;
struct snd_pcm *pcm;
struct azx_pcm *apcm;
int pcm_dev = cpcm->device;
int s, err;
/* if no substreams are defined for both playback and capture,
* it's just a placeholder. ignore it.
*/
if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
return 0;
if (snd_BUG_ON(!cpcm->name))
if (pcm_dev >= AZX_MAX_PCMS) {
snd_printk(KERN_ERR SFX "Invalid PCM device number %d\n",
pcm_dev);
return -EINVAL;
err = snd_pcm_new(chip->card, cpcm->name, cpcm->device,
cpcm->stream[0].substreams,
cpcm->stream[1].substreams,
}
if (chip->pcm[pcm_dev]) {
snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev);
return -EBUSY;
}
err = snd_pcm_new(chip->card, cpcm->name, pcm_dev,
cpcm->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams,
cpcm->stream[SNDRV_PCM_STREAM_CAPTURE].substreams,
&pcm);
if (err < 0)
return err;
strcpy(pcm->name, cpcm->name);
apcm = kmalloc(sizeof(*apcm), GFP_KERNEL);
apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
if (apcm == NULL)
return -ENOMEM;
apcm->chip = chip;
apcm->codec = codec;
apcm->hinfo[0] = &cpcm->stream[0];
apcm->hinfo[1] = &cpcm->stream[1];
pcm->private_data = apcm;
pcm->private_free = azx_pcm_free;
if (cpcm->stream[0].substreams)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops);
if (cpcm->stream[1].substreams)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops);
if (cpcm->pcm_type == HDA_PCM_TYPE_MODEM)
pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
chip->pcm[pcm_dev] = pcm;
cpcm->pcm = pcm;
for (s = 0; s < 2; s++) {
apcm->hinfo[s] = &cpcm->stream[s];
if (cpcm->stream[s].substreams)
snd_pcm_set_ops(pcm, s, &azx_pcm_ops);
}
/* buffer pre-allocation */
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
snd_dma_pci_data(chip->pci),
1024 * 64, 32 * 1024 * 1024);
chip->pcm[cpcm->device] = pcm;
return 0;
}
static int __devinit azx_pcm_create(struct azx *chip)
{
static const char *dev_name[HDA_PCM_NTYPES] = {
"Audio", "SPDIF", "HDMI", "Modem"
};
/* starting device index for each PCM type */
static int dev_idx[HDA_PCM_NTYPES] = {
[HDA_PCM_TYPE_AUDIO] = 0,
[HDA_PCM_TYPE_SPDIF] = 1,
[HDA_PCM_TYPE_HDMI] = 3,
[HDA_PCM_TYPE_MODEM] = 6
};
/* normal audio device indices; not linear to keep compatibility */
static int audio_idx[4] = { 0, 2, 4, 5 };
struct hda_codec *codec;
int c, err;
int num_devs[HDA_PCM_NTYPES];
err = snd_hda_build_pcms(chip->bus);
if (err < 0)
return err;
/* create audio PCMs */
memset(num_devs, 0, sizeof(num_devs));
list_for_each_entry(codec, &chip->bus->codec_list, list) {
for (c = 0; c < codec->num_pcms; c++) {
struct hda_pcm *cpcm = &codec->pcm_info[c];
int type = cpcm->pcm_type;
switch (type) {
case HDA_PCM_TYPE_AUDIO:
if (num_devs[type] >= ARRAY_SIZE(audio_idx)) {
snd_printk(KERN_WARNING
"Too many audio devices\n");
continue;
}
cpcm->device = audio_idx[num_devs[type]];
break;
case HDA_PCM_TYPE_SPDIF:
case HDA_PCM_TYPE_HDMI:
case HDA_PCM_TYPE_MODEM:
if (num_devs[type]) {
snd_printk(KERN_WARNING
"%s already defined\n",
dev_name[type]);
continue;
}
cpcm->device = dev_idx[type];
break;
default:
snd_printk(KERN_WARNING
"Invalid PCM type %d\n", type);
continue;
}
num_devs[type]++;
err = create_codec_pcm(chip, codec, cpcm);
if (err < 0)
return err;
}
}
return 0;
}
@ -1903,13 +1887,13 @@ static void azx_stop_chip(struct azx *chip)
#ifdef CONFIG_SND_HDA_POWER_SAVE
/* power-up/down the controller */
static void azx_power_notify(struct hda_codec *codec)
static void azx_power_notify(struct hda_bus *bus)
{
struct azx *chip = codec->bus->private_data;
struct azx *chip = bus->private_data;
struct hda_codec *c;
int power_on = 0;
list_for_each_entry(c, &codec->bus->codec_list, list) {
list_for_each_entry(c, &bus->codec_list, list) {
if (c->power_on) {
power_on = 1;
break;
@ -1926,6 +1910,18 @@ static void azx_power_notify(struct hda_codec *codec)
/*
* power management
*/
static int snd_hda_codecs_inuse(struct hda_bus *bus)
{
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
if (snd_hda_codec_needs_resume(codec))
return 1;
}
return 0;
}
static int azx_suspend(struct pci_dev *pci, pm_message_t state)
{
struct snd_card *card = pci_get_drvdata(pci);
@ -1951,13 +1947,16 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
return 0;
}
static int azx_resume_early(struct pci_dev *pci)
{
return pci_restore_state(pci);
}
static int azx_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct azx *chip = card->private_data;
pci_set_power_state(pci, PCI_D0);
pci_restore_state(pci);
if (pci_enable_device(pci) < 0) {
printk(KERN_ERR "hda-intel: pci_enable_device failed, "
"disabling device\n");
@ -2095,6 +2094,10 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
SND_PCI_QUIRK(0x1014, 0x05b7, "Thinkpad Z60", 0x01),
SND_PCI_QUIRK(0x17aa, 0x2010, "Thinkpad X/T/R60", 0x01),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X/T/R61", 0x01),
/* broken BIOS */
SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
{}
};
@ -2229,6 +2232,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->playback_streams = ATIHDMI_NUM_PLAYBACK;
chip->capture_streams = ATIHDMI_NUM_CAPTURE;
break;
case AZX_DRIVER_GENERIC:
default:
chip->playback_streams = ICH6_NUM_PLAYBACK;
chip->capture_streams = ICH6_NUM_CAPTURE;
@ -2338,40 +2342,31 @@ static int __devinit azx_probe(struct pci_dev *pci,
}
err = azx_create(card, pci, dev, pci_id->driver_data, &chip);
if (err < 0) {
snd_card_free(card);
return err;
}
if (err < 0)
goto out_free;
card->private_data = chip;
/* create codec instances */
err = azx_codec_create(chip, model[dev], probe_mask[dev]);
if (err < 0) {
snd_card_free(card);
return err;
}
err = azx_codec_create(chip, model[dev], probe_mask[dev],
probe_only[dev]);
if (err < 0)
goto out_free;
/* create PCM streams */
err = azx_pcm_create(chip);
if (err < 0) {
snd_card_free(card);
return err;
}
err = snd_hda_build_pcms(chip->bus);
if (err < 0)
goto out_free;
/* create mixer controls */
err = azx_mixer_create(chip);
if (err < 0) {
snd_card_free(card);
return err;
}
if (err < 0)
goto out_free;
snd_card_set_dev(card, &pci->dev);
err = snd_card_register(card);
if (err < 0) {
snd_card_free(card);
return err;
}
if (err < 0)
goto out_free;
pci_set_drvdata(pci, card);
chip->running = 1;
@ -2380,6 +2375,9 @@ static int __devinit azx_probe(struct pci_dev *pci,
dev++;
return err;
out_free:
snd_card_free(card);
return err;
}
static void __devexit azx_remove(struct pci_dev *pci)
@ -2453,6 +2451,11 @@ static struct pci_device_id azx_ids[] = {
{ PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA },
/* Teradici */
{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
/* AMD Generic, PCI class code and Vendor ID for HD Audio */
{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
.class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
.class_mask = 0xffffff,
.driver_data = AZX_DRIVER_GENERIC },
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
@ -2465,6 +2468,7 @@ static struct pci_driver driver = {
.remove = __devexit_p(azx_remove),
#ifdef CONFIG_PM
.suspend = azx_suspend,
.resume_early = azx_resume_early,
.resume = azx_resume,
#endif
};

100
sound/pci/hda/hda_local.h
View File

@ -96,6 +96,8 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
const char *name);
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char **slaves);
void snd_hda_codec_reset(struct hda_codec *codec);
int snd_hda_codec_configure(struct hda_codec *codec);
/* amp value bits */
#define HDA_AMP_MUTE 0x80
@ -282,6 +284,12 @@ int snd_hda_codec_proc_new(struct hda_codec *codec);
static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
#endif
#define SND_PRINT_RATES_ADVISED_BUFSIZE 80
void snd_print_pcm_rates(int pcm, char *buf, int buflen);
#define SND_PRINT_BITS_ADVISED_BUFSIZE 16
void snd_print_pcm_bits(int pcm, char *buf, int buflen);
/*
* Misc
*/
@ -364,17 +372,17 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
/* amp values */
#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8))
#define AMP_OUT_MUTE 0xb080
#define AMP_OUT_UNMUTE 0xb000
#define AMP_OUT_ZERO 0xb000
#define AMP_OUT_MUTE 0xb080
#define AMP_OUT_UNMUTE 0xb000
#define AMP_OUT_ZERO 0xb000
/* pinctl values */
#define PIN_IN (AC_PINCTL_IN_EN)
#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ)
#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ)
#define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50)
#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD)
#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD)
#define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80)
#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100)
#define PIN_OUT (AC_PINCTL_OUT_EN)
#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100)
#define PIN_OUT (AC_PINCTL_OUT_EN)
#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN)
#define PIN_HP_AMP (AC_PINCTL_HP_EN)
@ -393,10 +401,26 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps);
int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl);
void snd_hda_ctls_clear(struct hda_codec *codec);
/*
* hwdep interface
*/
#ifdef CONFIG_SND_HDA_HWDEP
int snd_hda_create_hwdep(struct hda_codec *codec);
#else
static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; }
#endif
#ifdef CONFIG_SND_HDA_RECONFIG
int snd_hda_hwdep_add_sysfs(struct hda_codec *codec);
#else
static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
{
return 0;
}
#endif
/*
* power-management
@ -430,4 +454,66 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1)
#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf)
/*
* CEA Short Audio Descriptor data
*/
struct cea_sad {
int channels;
int format; /* (format == 0) indicates invalid SAD */
int rates;
int sample_bits; /* for LPCM */
int max_bitrate; /* for AC3...ATRAC */
int profile; /* for WMAPRO */
};
#define ELD_FIXED_BYTES 20
#define ELD_MAX_MNL 16
#define ELD_MAX_SAD 16
/*
* ELD: EDID Like Data
*/
struct hdmi_eld {
int eld_size;
int baseline_len;
int eld_ver; /* (eld_ver == 0) indicates invalid ELD */
int cea_edid_ver;
char monitor_name[ELD_MAX_MNL + 1];
int manufacture_id;
int product_id;
u64 port_id;
int support_hdcp;
int support_ai;
int conn_type;
int aud_synch_delay;
int spk_alloc;
int sad_count;
struct cea_sad sad[ELD_MAX_SAD];
#ifdef CONFIG_PROC_FS
struct snd_info_entry *proc_entry;
#endif
};
int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
void snd_hdmi_show_eld(struct hdmi_eld *eld);
#ifdef CONFIG_PROC_FS
int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld);
void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld);
#else
static inline int snd_hda_eld_proc_new(struct hda_codec *codec,
struct hdmi_eld *eld)
{
return 0;
}
static inline void snd_hda_eld_proc_free(struct hda_codec *codec,
struct hdmi_eld *eld)
{
}
#endif
#define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80
void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen);
#endif /* __SOUND_HDA_LOCAL_H */

22
sound/pci/hda/hda_patch.h
View File

@ -1,22 +0,0 @@
/*
* HDA Patches - included by hda_codec.c
*/
/* Realtek codecs */
extern struct hda_codec_preset snd_hda_preset_realtek[];
/* C-Media codecs */
extern struct hda_codec_preset snd_hda_preset_cmedia[];
/* Analog Devices codecs */
extern struct hda_codec_preset snd_hda_preset_analog[];
/* SigmaTel codecs */
extern struct hda_codec_preset snd_hda_preset_sigmatel[];
/* SiLabs 3054/3055 modem codecs */
extern struct hda_codec_preset snd_hda_preset_si3054[];
/* ATI HDMI codecs */
extern struct hda_codec_preset snd_hda_preset_atihdmi[];
/* Conexant audio codec */
extern struct hda_codec_preset snd_hda_preset_conexant[];
/* VIA codecs */
extern struct hda_codec_preset snd_hda_preset_via[];
/* NVIDIA HDMI codecs */
extern struct hda_codec_preset snd_hda_preset_nvhdmi[];

91
sound/pci/hda/hda_proc.c
View File

@ -91,31 +91,21 @@ static void print_amp_vals(struct snd_info_buffer *buffer,
static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm)
{
static unsigned int rates[] = {
8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
96000, 176400, 192000, 384000
};
int i;
char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
pcm &= AC_SUPPCM_RATES;
snd_iprintf(buffer, " rates [0x%x]:", pcm);
for (i = 0; i < ARRAY_SIZE(rates); i++)
if (pcm & (1 << i))
snd_iprintf(buffer, " %d", rates[i]);
snd_iprintf(buffer, "\n");
snd_print_pcm_rates(pcm, buf, sizeof(buf));
snd_iprintf(buffer, "%s\n", buf);
}
static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm)
{
static unsigned int bits[] = { 8, 16, 20, 24, 32 };
int i;
char buf[SND_PRINT_BITS_ADVISED_BUFSIZE];
pcm = (pcm >> 16) & 0xff;
snd_iprintf(buffer, " bits [0x%x]:", pcm);
for (i = 0; i < ARRAY_SIZE(bits); i++)
if (pcm & (1 << i))
snd_iprintf(buffer, " %d", bits[i]);
snd_iprintf(buffer, "\n");
snd_iprintf(buffer, " bits [0x%x]:", (pcm >> 16) & 0xff);
snd_print_pcm_bits(pcm, buf, sizeof(buf));
snd_iprintf(buffer, "%s\n", buf);
}
static void print_pcm_formats(struct snd_info_buffer *buffer,
@ -145,32 +135,6 @@ static void print_pcm_caps(struct snd_info_buffer *buffer,
print_pcm_formats(buffer, stream);
}
static const char *get_jack_location(u32 cfg)
{
static char *bases[7] = {
"N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom",
};
static unsigned char specials_idx[] = {
0x07, 0x08,
0x17, 0x18, 0x19,
0x37, 0x38
};
static char *specials[] = {
"Rear Panel", "Drive Bar",
"Riser", "HDMI", "ATAPI",
"Mobile-In", "Mobile-Out"
};
int i;
cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT;
if ((cfg & 0x0f) < 7)
return bases[cfg & 0x0f];
for (i = 0; i < ARRAY_SIZE(specials_idx); i++) {
if (cfg == specials_idx[i])
return specials[i];
}
return "UNKNOWN";
}
static const char *get_jack_connection(u32 cfg)
{
static char *names[16] = {
@ -206,13 +170,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
int *supports_vref)
{
static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" };
static char *jack_types[16] = {
"Line Out", "Speaker", "HP Out", "CD",
"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
"Line In", "Aux", "Mic", "Telephony",
"SPDIF In", "Digitial In", "Reserved", "Other"
};
static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" };
unsigned int caps, val;
caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
@ -274,9 +231,9 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps,
jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT],
jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT],
jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3],
get_jack_location(caps));
snd_hda_get_jack_type(caps),
snd_hda_get_jack_connectivity(caps),
snd_hda_get_jack_location(caps));
snd_iprintf(buffer, " Conn = %s, Color = %s\n",
get_jack_connection(caps),
get_jack_color(caps));
@ -457,17 +414,6 @@ static void print_conn_list(struct snd_info_buffer *buffer,
}
}
static void print_realtek_coef(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid)
{
int coeff = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PROC_COEF, 0);
snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff);
coeff = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_COEF_INDEX, 0);
snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff);
}
static void print_gpio(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid)
{
@ -500,12 +446,13 @@ static void print_gpio(struct snd_info_buffer *buffer,
for (i = 0; i < max; ++i)
snd_iprintf(buffer,
" IO[%d]: enable=%d, dir=%d, wake=%d, "
"sticky=%d, data=%d\n", i,
"sticky=%d, data=%d, unsol=%d\n", i,
(enable & (1<<i)) ? 1 : 0,
(direction & (1<<i)) ? 1 : 0,
(wake & (1<<i)) ? 1 : 0,
(sticky & (1<<i)) ? 1 : 0,
(data & (1<<i)) ? 1 : 0);
(data & (1<<i)) ? 1 : 0,
(unsol & (1<<i)) ? 1 : 0);
/* FIXME: add GPO and GPI pin information */
}
@ -513,12 +460,11 @@ static void print_codec_info(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct hda_codec *codec = entry->private_data;
char buf[32];
hda_nid_t nid;
int i, nodes;
snd_hda_get_codec_name(codec, buf, sizeof(buf));
snd_iprintf(buffer, "Codec: %s\n", buf);
snd_iprintf(buffer, "Codec: %s\n",
codec->name ? codec->name : "Not Set");
snd_iprintf(buffer, "Address: %d\n", codec->addr);
snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id);
snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id);
@ -547,6 +493,8 @@ static void print_codec_info(struct snd_info_entry *entry,
}
print_gpio(buffer, codec, codec->afg);
if (codec->proc_widget_hook)
codec->proc_widget_hook(buffer, codec, codec->afg);
for (i = 0; i < nodes; i++, nid++) {
unsigned int wid_caps =
@ -649,9 +597,8 @@ static void print_codec_info(struct snd_info_entry *entry,
if (wid_caps & AC_WCAP_PROC_WID)
print_proc_caps(buffer, codec, nid);
/* NID 0x20 == Realtek Define Registers */
if (codec->vendor_id == 0x10ec && nid == 0x20)
print_realtek_coef(buffer, codec, nid);
if (codec->proc_widget_hook)
codec->proc_widget_hook(buffer, codec, nid);
}
snd_hda_power_down(codec);
}

132
sound/pci/hda/patch_analog.c
View File

@ -27,7 +27,6 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_patch.h"
struct ad198x_spec {
struct snd_kcontrol_new *mixers[5];
@ -67,8 +66,7 @@ struct ad198x_spec {
/* dynamic controls, init_verbs and input_mux */
struct auto_pin_cfg autocfg;
unsigned int num_kctl_alloc, num_kctl_used;
struct snd_kcontrol_new *kctl_alloc;
struct snd_array kctls;
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
@ -154,6 +152,8 @@ static const char *ad_slave_sws[] = {
NULL
};
static void ad198x_free_kctls(struct hda_codec *codec);
static int ad198x_build_controls(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
@ -202,6 +202,7 @@ static int ad198x_build_controls(struct hda_codec *codec)
return err;
}
ad198x_free_kctls(codec); /* no longer needed */
return 0;
}
@ -375,16 +376,27 @@ static int ad198x_build_pcms(struct hda_codec *codec)
return 0;
}
static void ad198x_free_kctls(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
if (spec->kctls.list) {
struct snd_kcontrol_new *kctl = spec->kctls.list;
int i;
for (i = 0; i < spec->kctls.used; i++)
kfree(kctl[i].name);
}
snd_array_free(&spec->kctls);
}
static void ad198x_free(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
unsigned int i;
if (spec->kctl_alloc) {
for (i = 0; i < spec->num_kctl_used; i++)
kfree(spec->kctl_alloc[i].name);
kfree(spec->kctl_alloc);
}
if (!spec)
return;
ad198x_free_kctls(codec);
kfree(codec->spec);
}
@ -625,6 +637,36 @@ static struct hda_input_mux ad1986a_automic_capture_source = {
};
static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.info = ad198x_mux_enum_info,
.get = ad198x_mux_enum_get,
.put = ad198x_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "External Amplifier",
.info = ad198x_eapd_info,
.get = ad198x_eapd_get,
.put = ad198x_eapd_put,
.private_value = 0x1b | (1 << 8), /* port-D, inversed */
},
{ } /* end */
};
static struct snd_kcontrol_new ad1986a_samsung_mixers[] = {
HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@ -917,6 +959,7 @@ enum {
AD1986A_LAPTOP_EAPD,
AD1986A_LAPTOP_AUTOMUTE,
AD1986A_ULTRA,
AD1986A_SAMSUNG,
AD1986A_MODELS
};
@ -927,6 +970,7 @@ static const char *ad1986a_models[AD1986A_MODELS] = {
[AD1986A_LAPTOP_EAPD] = "laptop-eapd",
[AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
[AD1986A_ULTRA] = "ultra",
[AD1986A_SAMSUNG] = "samsung",
};
static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
@ -949,9 +993,9 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD),
SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_LAPTOP_EAPD),
SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_LAPTOP_EAPD),
SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_LAPTOP_EAPD),
SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG),
SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG),
SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG),
SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
@ -1033,6 +1077,17 @@ static int patch_ad1986a(struct hda_codec *codec)
break;
case AD1986A_LAPTOP_EAPD:
spec->mixers[0] = ad1986a_laptop_eapd_mixers;
spec->num_init_verbs = 2;
spec->init_verbs[1] = ad1986a_eapd_init_verbs;
spec->multiout.max_channels = 2;
spec->multiout.num_dacs = 1;
spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
if (!is_jack_available(codec, 0x25))
spec->multiout.dig_out_nid = 0;
spec->input_mux = &ad1986a_laptop_eapd_capture_source;
break;
case AD1986A_SAMSUNG:
spec->mixers[0] = ad1986a_samsung_mixers;
spec->num_init_verbs = 3;
spec->init_verbs[1] = ad1986a_eapd_init_verbs;
spec->init_verbs[2] = ad1986a_automic_verbs;
@ -2452,9 +2507,6 @@ static struct hda_amp_list ad1988_loopbacks[] = {
* Automatic parse of I/O pins from the BIOS configuration
*/
#define NUM_CONTROL_ALLOC 32
#define NUM_VERB_ALLOC 32
enum {
AD_CTL_WIDGET_VOL,
AD_CTL_WIDGET_MUTE,
@ -2472,27 +2524,15 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name,
{
struct snd_kcontrol_new *knew;
if (spec->num_kctl_used >= spec->num_kctl_alloc) {
int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
if (! knew)
return -ENOMEM;
if (spec->kctl_alloc) {
memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
kfree(spec->kctl_alloc);
}
spec->kctl_alloc = knew;
spec->num_kctl_alloc = num;
}
knew = &spec->kctl_alloc[spec->num_kctl_used];
snd_array_init(&spec->kctls, sizeof(*knew), 32);
knew = snd_array_new(&spec->kctls);
if (!knew)
return -ENOMEM;
*knew = ad1988_control_templates[type];
knew->name = kstrdup(name, GFP_KERNEL);
if (! knew->name)
return -ENOMEM;
knew->private_value = val;
spec->num_kctl_used++;
return 0;
}
@ -2846,8 +2886,8 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = AD1988_SPDIF_IN;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
spec->init_verbs[spec->num_init_verbs++] = ad1988_6stack_init_verbs;
@ -3861,6 +3901,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
@ -4267,7 +4308,7 @@ static int patch_ad1882(struct hda_codec *codec)
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_analog[] = {
static struct hda_codec_preset snd_hda_preset_analog[] = {
{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
@ -4285,3 +4326,26 @@ struct hda_codec_preset snd_hda_preset_analog[] = {
{ .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:11d4*");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Analog Devices HD-audio codec");
static struct hda_codec_preset_list analog_list = {
.preset = snd_hda_preset_analog,
.owner = THIS_MODULE,
};
static int __init patch_analog_init(void)
{
return snd_hda_add_codec_preset(&analog_list);
}
static void __exit patch_analog_exit(void)
{
snd_hda_delete_codec_preset(&analog_list);
}
module_init(patch_analog_init)
module_exit(patch_analog_exit)

40
sound/pci/hda/patch_atihdmi.c
View File

@ -27,7 +27,6 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_patch.h"
struct atihdmi_spec {
struct hda_multi_out multiout;
@ -187,13 +186,40 @@ static int patch_atihdmi(struct hda_codec *codec)
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_atihdmi[] = {
{ .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
{ .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
{ .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi },
{ .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi },
static struct hda_codec_preset snd_hda_preset_atihdmi[] = {
{ .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi },
{ .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi },
{ .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi },
{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi },
{ .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi },
{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi },
{ .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:1002793c");
MODULE_ALIAS("snd-hda-codec-id:10027919");
MODULE_ALIAS("snd-hda-codec-id:1002791a");
MODULE_ALIAS("snd-hda-codec-id:1002aa01");
MODULE_ALIAS("snd-hda-codec-id:10951390");
MODULE_ALIAS("snd-hda-codec-id:17e80047");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("ATI HDMI HD-audio codec");
static struct hda_codec_preset_list atihdmi_list = {
.preset = snd_hda_preset_atihdmi,
.owner = THIS_MODULE,
};
static int __init patch_atihdmi_init(void)
{
return snd_hda_add_codec_preset(&atihdmi_list);
}
static void __exit patch_atihdmi_exit(void)
{
snd_hda_delete_codec_preset(&atihdmi_list);
}
module_init(patch_atihdmi_init)
module_exit(patch_atihdmi_exit)

27
sound/pci/hda/patch_cmedia.c
View File

@ -28,7 +28,6 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_patch.h"
#define NUM_PINS 11
@ -736,8 +735,32 @@ static int patch_cmi9880(struct hda_codec *codec)
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_cmedia[] = {
static struct hda_codec_preset snd_hda_preset_cmedia[] = {
{ .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
{ .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:13f69880");
MODULE_ALIAS("snd-hda-codec-id:434d4980");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("C-Media HD-audio codec");
static struct hda_codec_preset_list cmedia_list = {
.preset = snd_hda_preset_cmedia,
.owner = THIS_MODULE,
};
static int __init patch_cmedia_init(void)
{
return snd_hda_add_codec_preset(&cmedia_list);
}
static void __exit patch_cmedia_exit(void)
{
snd_hda_delete_codec_preset(&cmedia_list);
}
module_init(patch_cmedia_init)
module_exit(patch_cmedia_exit)

39
sound/pci/hda/patch_conexant.c
View File

@ -27,7 +27,6 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_patch.h"
#define CXT_PIN_DIR_IN 0x00
#define CXT_PIN_DIR_OUT 0x01
@ -86,8 +85,6 @@ struct conexant_spec {
/* dynamic controls, init_verbs and input_mux */
struct auto_pin_cfg autocfg;
unsigned int num_kctl_alloc, num_kctl_used;
struct snd_kcontrol_new *kctl_alloc;
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
@ -344,15 +341,6 @@ static int conexant_init(struct hda_codec *codec)
static void conexant_free(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
unsigned int i;
if (spec->kctl_alloc) {
for (i = 0; i < spec->num_kctl_used; i++)
kfree(spec->kctl_alloc[i].name);
kfree(spec->kctl_alloc);
}
kfree(codec->spec);
}
@ -1782,7 +1770,7 @@ static int patch_cxt5051(struct hda_codec *codec)
/*
*/
struct hda_codec_preset snd_hda_preset_conexant[] = {
static struct hda_codec_preset snd_hda_preset_conexant[] = {
{ .id = 0x14f15045, .name = "CX20549 (Venice)",
.patch = patch_cxt5045 },
{ .id = 0x14f15047, .name = "CX20551 (Waikiki)",
@ -1791,3 +1779,28 @@ struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_cxt5051 },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:14f15045");
MODULE_ALIAS("snd-hda-codec-id:14f15047");
MODULE_ALIAS("snd-hda-codec-id:14f15051");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
static struct hda_codec_preset_list conexant_list = {
.preset = snd_hda_preset_conexant,
.owner = THIS_MODULE,
};
static int __init patch_conexant_init(void)
{
return snd_hda_add_codec_preset(&conexant_list);
}
static void __exit patch_conexant_exit(void)
{
snd_hda_delete_codec_preset(&conexant_list);
}
module_init(patch_conexant_init)
module_exit(patch_conexant_exit)

711
sound/pci/hda/patch_intelhdmi.c Normal file
View File

@ -0,0 +1,711 @@
/*
*
* patch_intelhdmi.c - Patch for Intel HDMI codecs
*
* Copyright(c) 2008 Intel Corporation. All rights reserved.
*
* Authors:
* Jiang Zhe <zhe.jiang@intel.com>
* Wu Fengguang <wfg@linux.intel.com>
*
* Maintained by:
* Wu Fengguang <wfg@linux.intel.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the Free
* Software Foundation; either version 2 of the License, or (at your option)
* any later version.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY
* or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
* for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software Foundation,
* Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#define CVT_NID 0x02 /* audio converter */
#define PIN_NID 0x03 /* HDMI output pin */
#define INTEL_HDMI_EVENT_TAG 0x08
struct intel_hdmi_spec {
struct hda_multi_out multiout;
struct hda_pcm pcm_rec;
struct hdmi_eld sink_eld;
};
static struct hda_verb pinout_enable_verb[] = {
{PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{} /* terminator */
};
static struct hda_verb pinout_disable_verb[] = {
{PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00},
{}
};
static struct hda_verb unsolicited_response_verb[] = {
{PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN |
INTEL_HDMI_EVENT_TAG},
{}
};
static struct hda_verb def_chan_map[] = {
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00},
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11},
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22},
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33},
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44},
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55},
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66},
{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77},
{}
};
struct hdmi_audio_infoframe {
u8 type; /* 0x84 */
u8 ver; /* 0x01 */
u8 len; /* 0x0a */
u8 checksum; /* PB0 */
u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */
u8 SS01_SF24;
u8 CXT04;
u8 CA;
u8 LFEPBL01_LSV36_DM_INH7;
u8 reserved[5]; /* PB6 - PB10 */
};
/*
* CEA speaker placement:
*
* FLH FCH FRH
* FLW FL FLC FC FRC FR FRW
*
* LFE
* TC
*
* RL RLC RC RRC RR
*
* The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to
* CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC.
*/
enum cea_speaker_placement {
FL = (1 << 0), /* Front Left */
FC = (1 << 1), /* Front Center */
FR = (1 << 2), /* Front Right */
FLC = (1 << 3), /* Front Left Center */
FRC = (1 << 4), /* Front Right Center */
RL = (1 << 5), /* Rear Left */
RC = (1 << 6), /* Rear Center */
RR = (1 << 7), /* Rear Right */
RLC = (1 << 8), /* Rear Left Center */
RRC = (1 << 9), /* Rear Right Center */
LFE = (1 << 10), /* Low Frequency Effect */
FLW = (1 << 11), /* Front Left Wide */
FRW = (1 << 12), /* Front Right Wide */
FLH = (1 << 13), /* Front Left High */
FCH = (1 << 14), /* Front Center High */
FRH = (1 << 15), /* Front Right High */
TC = (1 << 16), /* Top Center */
};
/*
* ELD SA bits in the CEA Speaker Allocation data block
*/
static int eld_speaker_allocation_bits[] = {
[0] = FL | FR,
[1] = LFE,
[2] = FC,
[3] = RL | RR,
[4] = RC,
[5] = FLC | FRC,
[6] = RLC | RRC,
/* the following are not defined in ELD yet */
[7] = FLW | FRW,
[8] = FLH | FRH,
[9] = TC,
[10] = FCH,
};
struct cea_channel_speaker_allocation {
int ca_index;
int speakers[8];
/* derived values, just for convenience */
int channels;
int spk_mask;
};
/*
* This is an ordered list!
*
* The preceding ones have better chances to be selected by
* hdmi_setup_channel_allocation().
*/
static struct cea_channel_speaker_allocation channel_allocations[] = {
/* channel: 8 7 6 5 4 3 2 1 */
{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } },
/* 2.1 */
{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } },
/* Dolby Surround */
{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } },
{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } },
{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } },
{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } },
{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } },
{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } },
{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } },
/* 5.1 */
{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } },
/* 6.1 */
{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } },
/* 7.1 */
{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } },
{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } },
{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } },
{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } },
{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } },
{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } },
{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } },
{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } },
{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } },
{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } },
{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } },
{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } },
{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } },
};
/*
* HDMI routines
*/
#ifdef BE_PARANOID
static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t nid,
int *packet_index, int *byte_index)
{
int val;
val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_INDEX, 0);
*packet_index = val >> 5;
*byte_index = val & 0x1f;
}
#endif
static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t nid,
int packet_index, int byte_index)
{
int val;
val = (packet_index << 5) | (byte_index & 0x1f);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val);
}
static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
unsigned char val)
{
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val);
}
static void hdmi_enable_output(struct hda_codec *codec)
{
/* Enable Audio InfoFrame Transmission */
hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
AC_DIPXMIT_BEST);
/* Unmute */
if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, PIN_NID, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
/* Enable pin out */
snd_hda_sequence_write(codec, pinout_enable_verb);
}
static void hdmi_disable_output(struct hda_codec *codec)
{
snd_hda_sequence_write(codec, pinout_disable_verb);
if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, PIN_NID, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
/*
* FIXME: noises may arise when playing music after reloading the
* kernel module, until the next X restart or monitor repower.
*/
}
static int hdmi_get_channel_count(struct hda_codec *codec)
{
return 1 + snd_hda_codec_read(codec, CVT_NID, 0,
AC_VERB_GET_CVT_CHAN_COUNT, 0);
}
static void hdmi_set_channel_count(struct hda_codec *codec, int chs)
{
snd_hda_codec_write(codec, CVT_NID, 0,
AC_VERB_SET_CVT_CHAN_COUNT, chs - 1);
if (chs != hdmi_get_channel_count(codec))
snd_printd(KERN_INFO "HDMI channel count: expect %d, get %d\n",
chs, hdmi_get_channel_count(codec));
}
static void hdmi_debug_channel_mapping(struct hda_codec *codec)
{
#ifdef CONFIG_SND_DEBUG_VERBOSE
int i;
int slot;
for (i = 0; i < 8; i++) {
slot = snd_hda_codec_read(codec, CVT_NID, 0,
AC_VERB_GET_HDMI_CHAN_SLOT, i);
printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n",
slot >> 4, slot & 0x7);
}
#endif
}
static void hdmi_parse_eld(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->sink_eld;
if (!snd_hdmi_get_eld(eld, codec, PIN_NID))
snd_hdmi_show_eld(eld);
}
/*
* Audio InfoFrame routines
*/
static void hdmi_debug_dip_size(struct hda_codec *codec)
{
#ifdef CONFIG_SND_DEBUG_VERBOSE
int i;
int size;
size = snd_hdmi_get_eld_size(codec, PIN_NID);
printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size);
for (i = 0; i < 8; i++) {
size = snd_hda_codec_read(codec, PIN_NID, 0,
AC_VERB_GET_HDMI_DIP_SIZE, i);
printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size);
}
#endif
}
static void hdmi_clear_dip_buffers(struct hda_codec *codec)
{
#ifdef BE_PARANOID
int i, j;
int size;
int pi, bi;
for (i = 0; i < 8; i++) {
size = snd_hda_codec_read(codec, PIN_NID, 0,
AC_VERB_GET_HDMI_DIP_SIZE, i);
if (size == 0)
continue;
hdmi_set_dip_index(codec, PIN_NID, i, 0x0);
for (j = 1; j < 1000; j++) {
hdmi_write_dip_byte(codec, PIN_NID, 0x0);
hdmi_get_dip_index(codec, PIN_NID, &pi, &bi);
if (pi != i)
snd_printd(KERN_INFO "dip index %d: %d != %d\n",
bi, pi, i);
if (bi == 0) /* byte index wrapped around */
break;
}
snd_printd(KERN_INFO
"HDMI: DIP GP[%d] buf reported size=%d, written=%d\n",
i, size, j);
}
#endif
}
static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
struct hdmi_audio_infoframe *ai)
{
u8 *params = (u8 *)ai;
int i;
hdmi_debug_dip_size(codec);
hdmi_clear_dip_buffers(codec); /* be paranoid */
hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
for (i = 0; i < sizeof(ai); i++)
hdmi_write_dip_byte(codec, PIN_NID, params[i]);
}
/*
* Compute derived values in channel_allocations[].
*/
static void init_channel_allocations(void)
{
int i, j;
struct cea_channel_speaker_allocation *p;
for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
p = channel_allocations + i;
p->channels = 0;
p->spk_mask = 0;
for (j = 0; j < ARRAY_SIZE(p->speakers); j++)
if (p->speakers[j]) {
p->channels++;
p->spk_mask |= p->speakers[j];
}
}
}
/*
* The transformation takes two steps:
*
* eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask
* spk_mask => (channel_allocations[]) => ai->CA
*
* TODO: it could select the wrong CA from multiple candidates.
*/
static int hdmi_setup_channel_allocation(struct hda_codec *codec,
struct hdmi_audio_infoframe *ai)
{
struct intel_hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->sink_eld;
int i;
int spk_mask = 0;
int channels = 1 + (ai->CC02_CT47 & 0x7);
char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
/*
* CA defaults to 0 for basic stereo audio
*/
if (!eld->eld_ver)
return 0;
if (!eld->spk_alloc)
return 0;
if (channels <= 2)
return 0;
/*
* expand ELD's speaker allocation mask
*
* ELD tells the speaker mask in a compact(paired) form,
* expand ELD's notions to match the ones used by Audio InfoFrame.
*/
for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) {
if (eld->spk_alloc & (1 << i))
spk_mask |= eld_speaker_allocation_bits[i];
}
/* search for the first working match in the CA table */
for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
if (channels == channel_allocations[i].channels &&
(spk_mask & channel_allocations[i].spk_mask) ==
channel_allocations[i].spk_mask) {
ai->CA = channel_allocations[i].ca_index;
break;
}
}
snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf));
snd_printdd(KERN_INFO
"HDMI: select CA 0x%x for %d-channel allocation: %s\n",
ai->CA, channels, buf);
return ai->CA;
}
static void hdmi_setup_channel_mapping(struct hda_codec *codec,
struct hdmi_audio_infoframe *ai)
{
if (!ai->CA)
return;
/*
* TODO: adjust channel mapping if necessary
* ALSA sequence is front/surr/clfe/side?
*/
snd_hda_sequence_write(codec, def_chan_map);
hdmi_debug_channel_mapping(codec);
}
static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct hdmi_audio_infoframe ai = {
.type = 0x84,
.ver = 0x01,
.len = 0x0a,
.CC02_CT47 = substream->runtime->channels - 1,
};
hdmi_setup_channel_allocation(codec, &ai);
hdmi_setup_channel_mapping(codec, &ai);
hdmi_fill_audio_infoframe(codec, &ai);
}
/*
* Unsolicited events
*/
static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
{
int pind = !!(res & AC_UNSOL_RES_PD);
int eldv = !!(res & AC_UNSOL_RES_ELDV);
printk(KERN_INFO
"HDMI hot plug event: Presence_Detect=%d ELD_Valid=%d\n",
pind, eldv);
if (pind && eldv) {
hdmi_parse_eld(codec);
/* TODO: do real things about ELD */
}
}
static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
{
int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
int cp_state = !!(res & AC_UNSOL_RES_CP_STATE);
int cp_ready = !!(res & AC_UNSOL_RES_CP_READY);
printk(KERN_INFO
"HDMI content protection event: SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
subtag,
cp_state,
cp_ready);
/* TODO */
if (cp_state)
;
if (cp_ready)
;
}
static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
{
int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
if (tag != INTEL_HDMI_EVENT_TAG) {
snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag);
return;
}
if (subtag == 0)
hdmi_intrinsic_event(codec, res);
else
hdmi_non_intrinsic_event(codec, res);
}
/*
* Callbacks
*/
static int intel_hdmi_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct intel_hdmi_spec *spec = codec->spec;
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct intel_hdmi_spec *spec = codec->spec;
hdmi_disable_output(codec);
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct intel_hdmi_spec *spec = codec->spec;
snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
format, substream);
hdmi_set_channel_count(codec, substream->runtime->channels);
hdmi_setup_audio_infoframe(codec, substream);
hdmi_enable_output(codec);
return 0;
}
static struct hda_pcm_stream intel_hdmi_pcm_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
.nid = CVT_NID, /* NID to query formats and rates and setup streams */
.ops = {
.open = intel_hdmi_playback_pcm_open,
.close = intel_hdmi_playback_pcm_close,
.prepare = intel_hdmi_playback_pcm_prepare
},
};
static int intel_hdmi_build_pcms(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec = codec->spec;
struct hda_pcm *info = &spec->pcm_rec;
codec->num_pcms = 1;
codec->pcm_info = info;
info->name = "INTEL HDMI";
info->pcm_type = HDA_PCM_TYPE_HDMI;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback;
return 0;
}
static int intel_hdmi_build_controls(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec = codec->spec;
int err;
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
if (err < 0)
return err;
return 0;
}
static int intel_hdmi_init(struct hda_codec *codec)
{
/* disable audio output as early as possible */
hdmi_disable_output(codec);
snd_hda_sequence_write(codec, unsolicited_response_verb);
return 0;
}
static void intel_hdmi_free(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec = codec->spec;
snd_hda_eld_proc_free(codec, &spec->sink_eld);
kfree(spec);
}
static struct hda_codec_ops intel_hdmi_patch_ops = {
.init = intel_hdmi_init,
.free = intel_hdmi_free,
.build_pcms = intel_hdmi_build_pcms,
.build_controls = intel_hdmi_build_controls,
.unsol_event = intel_hdmi_unsol_event,
};
static int patch_intel_hdmi(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
spec->multiout.num_dacs = 0; /* no analog */
spec->multiout.max_channels = 8;
spec->multiout.dig_out_nid = CVT_NID;
codec->spec = spec;
codec->patch_ops = intel_hdmi_patch_ops;
snd_hda_eld_proc_new(codec, &spec->sink_eld);
init_channel_allocations();
return 0;
}
static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
{ .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi },
{ .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:808629fb");
MODULE_ALIAS("snd-hda-codec-id:80862801");
MODULE_ALIAS("snd-hda-codec-id:80862802");
MODULE_ALIAS("snd-hda-codec-id:80862803");
MODULE_ALIAS("snd-hda-codec-id:10951392");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Intel HDMI HD-audio codec");
static struct hda_codec_preset_list intel_list = {
.preset = snd_hda_preset_intelhdmi,
.owner = THIS_MODULE,
};
static int __init patch_intelhdmi_init(void)
{
return snd_hda_add_codec_preset(&intel_list);
}
static void __exit patch_intelhdmi_exit(void)
{
snd_hda_delete_codec_preset(&intel_list);
}
module_init(patch_intelhdmi_init)
module_exit(patch_intelhdmi_exit)

32
sound/pci/hda/patch_nvhdmi.c
View File

@ -158,8 +158,34 @@ static int patch_nvhdmi(struct hda_codec *codec)
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
{ .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi },
{ .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi },
static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi },
{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi },
{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:10de0002");
MODULE_ALIAS("snd-hda-codec-id:10de0007");
MODULE_ALIAS("snd-hda-codec-id:10de0067");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Nvidia HDMI HD-audio codec");
static struct hda_codec_preset_list nvhdmi_list = {
.preset = snd_hda_preset_nvhdmi,
.owner = THIS_MODULE,
};
static int __init patch_nvhdmi_init(void)
{
return snd_hda_add_codec_preset(&nvhdmi_list);
}
static void __exit patch_nvhdmi_exit(void)
{
snd_hda_delete_codec_preset(&nvhdmi_list);
}
module_init(patch_nvhdmi_init)
module_exit(patch_nvhdmi_exit)

1667
sound/pci/hda/patch_realtek.c
View File

File diff suppressed because it is too large Load Diff

35
sound/pci/hda/patch_si3054.c
View File

@ -28,7 +28,6 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_patch.h"
/* si3054 verbs */
#define SI3054_VERB_READ_NODE 0x900
@ -283,7 +282,7 @@ static int patch_si3054(struct hda_codec *codec)
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_si3054[] = {
static struct hda_codec_preset snd_hda_preset_si3054[] = {
{ .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
{ .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
{ .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
@ -301,3 +300,35 @@ struct hda_codec_preset snd_hda_preset_si3054[] = {
{}
};
MODULE_ALIAS("snd-hda-codec-id:163c3055");
MODULE_ALIAS("snd-hda-codec-id:163c3155");
MODULE_ALIAS("snd-hda-codec-id:11c13026");
MODULE_ALIAS("snd-hda-codec-id:11c13055");
MODULE_ALIAS("snd-hda-codec-id:11c13155");
MODULE_ALIAS("snd-hda-codec-id:10573055");
MODULE_ALIAS("snd-hda-codec-id:10573057");
MODULE_ALIAS("snd-hda-codec-id:10573155");
MODULE_ALIAS("snd-hda-codec-id:11063288");
MODULE_ALIAS("snd-hda-codec-id:15433155");
MODULE_ALIAS("snd-hda-codec-id:18540018");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Si3054 HD-audio modem codec");
static struct hda_codec_preset_list si3054_list = {
.preset = snd_hda_preset_si3054,
.owner = THIS_MODULE,
};
static int __init patch_si3054_init(void)
{
return snd_hda_add_codec_preset(&si3054_list);
}
static void __exit patch_si3054_exit(void)
{
snd_hda_delete_codec_preset(&si3054_list);
}
module_init(patch_si3054_init)
module_exit(patch_si3054_exit)

1326
sound/pci/hda/patch_sigmatel.c
View File

File diff suppressed because it is too large Load Diff

171
sound/pci/hda/patch_via.c
View File

@ -47,15 +47,11 @@
#include <sound/asoundef.h>
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_patch.h"
/* amp values */
#define AMP_VAL_IDX_SHIFT 19
#define AMP_VAL_IDX_MASK (0x0f<<19)
#define NUM_CONTROL_ALLOC 32
#define NUM_VERB_ALLOC 32
/* Pin Widget NID */
#define VT1708_HP_NID 0x13
#define VT1708_DIGOUT_NID 0x14
@ -145,8 +141,6 @@ enum {
AUTO_SEQ_SIDE
};
#define get_amp_nid(kc) ((kc)->private_value & 0xffff)
/* Some VT1708S based boards gets the micboost setting wrong, so we have
* to apply some brute-force and re-write the TLV's by software. */
static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,
@ -227,8 +221,7 @@ struct via_spec {
/* dynamic controls, init_verbs and input_mux */
struct auto_pin_cfg autocfg;
unsigned int num_kctl_alloc, num_kctl_used;
struct snd_kcontrol_new *kctl_alloc;
struct snd_array kctls;
struct hda_input_mux private_imux[2];
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
@ -272,33 +265,31 @@ static int via_add_control(struct via_spec *spec, int type, const char *name,
{
struct snd_kcontrol_new *knew;
if (spec->num_kctl_used >= spec->num_kctl_alloc) {
int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
/* array + terminator */
knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL);
if (!knew)
return -ENOMEM;
if (spec->kctl_alloc) {
memcpy(knew, spec->kctl_alloc,
sizeof(*knew) * spec->num_kctl_alloc);
kfree(spec->kctl_alloc);
}
spec->kctl_alloc = knew;
spec->num_kctl_alloc = num;
}
knew = &spec->kctl_alloc[spec->num_kctl_used];
snd_array_init(&spec->kctls, sizeof(*knew), 32);
knew = snd_array_new(&spec->kctls);
if (!knew)
return -ENOMEM;
*knew = vt1708_control_templates[type];
knew->name = kstrdup(name, GFP_KERNEL);
if (!knew->name)
return -ENOMEM;
knew->private_value = val;
spec->num_kctl_used++;
return 0;
}
static void via_free_kctls(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
if (spec->kctls.list) {
struct snd_kcontrol_new *kctl = spec->kctls.list;
int i;
for (i = 0; i < spec->kctls.used; i++)
kfree(kctl[i].name);
}
snd_array_free(&spec->kctls);
}
/* create input playback/capture controls for the given pin */
static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin,
const char *ctlname, int idx, int mix_nid)
@ -896,6 +887,7 @@ static int via_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
via_free_kctls(codec); /* no longer needed */
return 0;
}
@ -941,17 +933,11 @@ static int via_build_pcms(struct hda_codec *codec)
static void via_free(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
unsigned int i;
if (!spec)
return;
if (spec->kctl_alloc) {
for (i = 0; i < spec->num_kctl_used; i++)
kfree(spec->kctl_alloc[i].name);
kfree(spec->kctl_alloc);
}
via_free_kctls(codec);
kfree(codec->spec);
}
@ -1373,8 +1359,8 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = VT1708_DIGIN_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
spec->init_verbs[spec->num_iverbs++] = vt1708_volume_init_verbs;
@ -1846,8 +1832,8 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = VT1709_DIGIN_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
spec->input_mux = &spec->private_imux[0];
@ -2390,8 +2376,8 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = VT1708B_DIGIN_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
spec->input_mux = &spec->private_imux[0];
@ -2855,8 +2841,8 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
spec->extra_dig_out_nid = 0x15;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
spec->input_mux = &spec->private_imux[0];
@ -3174,8 +3160,8 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
spec->extra_dig_out_nid = 0x1B;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
spec->input_mux = &spec->private_imux[0];
@ -3262,74 +3248,97 @@ static int patch_vt1702(struct hda_codec *codec)
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_via[] = {
{ .id = 0x11061708, .name = "VIA VT1708", .patch = patch_vt1708},
{ .id = 0x11061709, .name = "VIA VT1708", .patch = patch_vt1708},
{ .id = 0x1106170A, .name = "VIA VT1708", .patch = patch_vt1708},
{ .id = 0x1106170B, .name = "VIA VT1708", .patch = patch_vt1708},
{ .id = 0x1106E710, .name = "VIA VT1709 10-Ch",
static struct hda_codec_preset snd_hda_preset_via[] = {
{ .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708},
{ .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708},
{ .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708},
{ .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708},
{ .id = 0x1106e710, .name = "VT1709 10-Ch",
.patch = patch_vt1709_10ch},
{ .id = 0x1106E711, .name = "VIA VT1709 10-Ch",
{ .id = 0x1106e711, .name = "VT1709 10-Ch",
.patch = patch_vt1709_10ch},
{ .id = 0x1106E712, .name = "VIA VT1709 10-Ch",
{ .id = 0x1106e712, .name = "VT1709 10-Ch",
.patch = patch_vt1709_10ch},
{ .id = 0x1106E713, .name = "VIA VT1709 10-Ch",
{ .id = 0x1106e713, .name = "VT1709 10-Ch",
.patch = patch_vt1709_10ch},
{ .id = 0x1106E714, .name = "VIA VT1709 6-Ch",
{ .id = 0x1106e714, .name = "VT1709 6-Ch",
.patch = patch_vt1709_6ch},
{ .id = 0x1106E715, .name = "VIA VT1709 6-Ch",
{ .id = 0x1106e715, .name = "VT1709 6-Ch",
.patch = patch_vt1709_6ch},
{ .id = 0x1106E716, .name = "VIA VT1709 6-Ch",
{ .id = 0x1106e716, .name = "VT1709 6-Ch",
.patch = patch_vt1709_6ch},
{ .id = 0x1106E717, .name = "VIA VT1709 6-Ch",
{ .id = 0x1106e717, .name = "VT1709 6-Ch",
.patch = patch_vt1709_6ch},
{ .id = 0x1106E720, .name = "VIA VT1708B 8-Ch",
{ .id = 0x1106e720, .name = "VT1708B 8-Ch",
.patch = patch_vt1708B_8ch},
{ .id = 0x1106E721, .name = "VIA VT1708B 8-Ch",
{ .id = 0x1106e721, .name = "VT1708B 8-Ch",
.patch = patch_vt1708B_8ch},
{ .id = 0x1106E722, .name = "VIA VT1708B 8-Ch",
{ .id = 0x1106e722, .name = "VT1708B 8-Ch",
.patch = patch_vt1708B_8ch},
{ .id = 0x1106E723, .name = "VIA VT1708B 8-Ch",
{ .id = 0x1106e723, .name = "VT1708B 8-Ch",
.patch = patch_vt1708B_8ch},
{ .id = 0x1106E724, .name = "VIA VT1708B 4-Ch",
{ .id = 0x1106e724, .name = "VT1708B 4-Ch",
.patch = patch_vt1708B_4ch},
{ .id = 0x1106E725, .name = "VIA VT1708B 4-Ch",
{ .id = 0x1106e725, .name = "VT1708B 4-Ch",
.patch = patch_vt1708B_4ch},
{ .id = 0x1106E726, .name = "VIA VT1708B 4-Ch",
{ .id = 0x1106e726, .name = "VT1708B 4-Ch",
.patch = patch_vt1708B_4ch},
{ .id = 0x1106E727, .name = "VIA VT1708B 4-Ch",
{ .id = 0x1106e727, .name = "VT1708B 4-Ch",
.patch = patch_vt1708B_4ch},
{ .id = 0x11060397, .name = "VIA VT1708S",
{ .id = 0x11060397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11061397, .name = "VIA VT1708S",
{ .id = 0x11061397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11062397, .name = "VIA VT1708S",
{ .id = 0x11062397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11063397, .name = "VIA VT1708S",
{ .id = 0x11063397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11064397, .name = "VIA VT1708S",
{ .id = 0x11064397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11065397, .name = "VIA VT1708S",
{ .id = 0x11065397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11066397, .name = "VIA VT1708S",
{ .id = 0x11066397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11067397, .name = "VIA VT1708S",
{ .id = 0x11067397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11060398, .name = "VIA VT1702",
{ .id = 0x11060398, .name = "VT1702",
.patch = patch_vt1702},
{ .id = 0x11061398, .name = "VIA VT1702",
{ .id = 0x11061398, .name = "VT1702",
.patch = patch_vt1702},
{ .id = 0x11062398, .name = "VIA VT1702",
{ .id = 0x11062398, .name = "VT1702",
.patch = patch_vt1702},
{ .id = 0x11063398, .name = "VIA VT1702",
{ .id = 0x11063398, .name = "VT1702",
.patch = patch_vt1702},
{ .id = 0x11064398, .name = "VIA VT1702",
{ .id = 0x11064398, .name = "VT1702",
.patch = patch_vt1702},
{ .id = 0x11065398, .name = "VIA VT1702",
{ .id = 0x11065398, .name = "VT1702",
.patch = patch_vt1702},
{ .id = 0x11066398, .name = "VIA VT1702",
{ .id = 0x11066398, .name = "VT1702",
.patch = patch_vt1702},
{ .id = 0x11067398, .name = "VIA VT1702",
{ .id = 0x11067398, .name = "VT1702",
.patch = patch_vt1702},
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:1106*");
static struct hda_codec_preset_list via_list = {
.preset = snd_hda_preset_via,
.owner = THIS_MODULE,
};
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("VIA HD-audio codec");
static int __init patch_via_init(void)
{
return snd_hda_add_codec_preset(&via_list);
}
static void __exit patch_via_exit(void)
{
snd_hda_delete_codec_preset(&via_list);
}
module_init(patch_via_init)
module_exit(patch_via_exit)

23
sound/pci/ice1712/ice1724.c
View File

@ -382,23 +382,25 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
unsigned char status_mask =
VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX | VT1724_IRQ_MTPCM;
int handled = 0;
#ifdef CONFIG_SND_DEBUG
int timeout = 0;
#endif
while (1) {
status = inb(ICEREG1724(ice, IRQSTAT));
status &= status_mask;
if (status == 0)
break;
#ifdef CONFIG_SND_DEBUG
if (++timeout > 10) {
printk(KERN_ERR
"ice1724: Too long irq loop, status = 0x%x\n",
status);
status = inb(ICEREG1724(ice, IRQSTAT));
printk(KERN_ERR "ice1724: Too long irq loop, "
"status = 0x%x\n", status);
if (status & VT1724_IRQ_MPU_TX) {
printk(KERN_ERR "ice1724: Disabling MPU_TX\n");
outb(inb(ICEREG1724(ice, IRQMASK)) |
VT1724_IRQ_MPU_TX,
ICEREG1724(ice, IRQMASK));
}
break;
}
#endif
handled = 1;
if (status & VT1724_IRQ_MPU_TX) {
spin_lock(&ice->reg_lock);
@ -2351,7 +2353,6 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
{
struct snd_ice1712 *ice;
int err;
unsigned char mask;
static struct snd_device_ops ops = {
.dev_free = snd_vt1724_dev_free,
};
@ -2412,9 +2413,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
return -EIO;
}
/* unmask used interrupts */
mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX;
outb(mask, ICEREG1724(ice, IRQMASK));
/* MPU_RX and TX irq masks are cleared later dynamically */
outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK));
/* don't handle FIFO overrun/underruns (just yet),
* since they cause machine lockups
*/

Some files were not shown because too many files have changed in this diff Show More