hollalinux
/
linux-sg2042
mirror of https://github.com/sophgo/linux-riscv.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity

Updates of sound stuff for 3.4-rc1 

Here is the first big update chunk of sound stuff for 3.4-rc1.
 
 In the common sound infrastructure, there are a few changes for
 dynamic PCM support (used in ASoC) and a few clean-ups.  Majority of
 changes are found, as usual, in HD-audio and ASoC.
 
 Some highlights of HD-audio changes:
 - All the long-standing static quirk codes for Realtek codec were
 finally removed by fixing and extending the Realtek auto-parser.
 
 - The mute-LED control is standardized over all HD-audio codec
   drivers using the extended vmaster hook.
 
 - The vmaster slave mixer elements are initialized to 0dB as default
   so that the user won't be annoyed by the silent output after
   updates, e.g. due to the additions of new elements.
 
 - Other many fix-ups for the misc HD-audio devices.
 
 In the ASoC side, this is a very active release, including a quite a
 few framework enhancements.  Some highlights:
 
 - Support for widgets not associated with a CODEC, an important part
   of the dynamic PCM framework.
 
 - A library factoring out the common code shared by dmaengine based
   DMA drivers contributed by Lars-Peter Clausen.  This will save a lot
   of code and make it much easier to deploy enhancements to
   dmaengine.
 
 - Support for binary controls, used for providing runtime
   configuration of algorithm coefficients.
 
 - A new DAPM widget type for regulator supplies allowing drivers for
   devices that can power down unused supplies while active to do
   without any per-driver code.
 
 - DAPM widgets for DAIs, initially giving a speed boost for playback
   startup and shutdown and also the basis for CODEC<->CODEC DAI link
   support.
 
 - Support for specifying the number of significant bits on audio
   interfaces, useful for allowing applications to know how much effort
   to put into generating data for a larger sample format.
 
 - Conversion of the FSI driver used on some SH processors to
   DMAEngine.
 
 - Conversion of EP93xx drivers to DMAEngine.
 
 - New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
   WM2200.
 
 - Move audmux driver from arc/arm to sound/soc
 
 - McBSP move from arch/ to sound/ and updates
 
 Also, a few small updates and fixes for other drivers like au88x0,
 ymfpci, USB 6fire, USB usx2yaudio are included.
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v2.0.18 (GNU/Linux)
 
 iQIcBAABAgAGBQJPawmuAAoJEGwxgFQ9KSmkdR0QALDDI/eUo4il40C33Gt/OSDz
 Wkz+FOwp2ddbIqHc0n41sV7+FV1MuyzQ37uSw8zhK724Vd4tCqX6O5K1uvS1iSbh
 jxHsy5XtfvCs2cjb61H6N+rfWcqC69gGhpc1mLoelj/PYl7S2iV5xOgTr4trVk2Y
 UN7Y13b4hzZvubRUozoTldaIgdhrj8D8KO7qQNYehyG19b4bJ00Rk4K5JdjsFwbE
 dTGl/ZUv50Fnx6PAWwqzh1a3cPabHA1TZDiKQM2nuE91e/Ecs4c7t1CRvW8m8mlr
 u4D4N8PJcCN4SPDd2YuVBgan4SQ0kxKTaup11bHSvAWai2zPX5xMB1yoJNxgjSMt
 5NHrGdR4+lQEVlBVXe2sWb4/3vE2kr2dtcGGR/FBFJTuLWDFFtRcnxeQJI8qRNUw
 UdwDuGXdActoc1cZz2dsKvXMOs0TKT6OCdQH+dHBglW/W8wMkVocZclUgbQM66/X
 gwvk0jfZ9p3UcKnYt3RkxiXQvAJsr8v0HhYcKvQCFhJArZufdeRHB7LCVRTm692Y
 /BKZgK4QHxtGw3Yc7emYidKeRSP1ml5QlvC4zMIoGqiahqa8LI8Qcb5knvIEmU8q
 kY5k0fVP+paf0dceAVXyFZsRB9AyX2eUdufDPifXtydQZgj4o9A7Sy/teWl77EgF
 Mafq4QUzo1U4i8JpAM4d
 =5FJq
 -----END PGP SIGNATURE-----

Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull updates of sound stuff from Takashi Iwai:
 "Here is the first big update chunk of sound stuff for 3.4-rc1.

  In the common sound infrastructure, there are a few changes for
  dynamic PCM support (used in ASoC) and a few clean-ups.  Majority of
  changes are found, as usual, in HD-audio and ASoC.

  Some highlights of HD-audio changes:

   - All the long-standing static quirk codes for Realtek codec were
     finally removed by fixing and extending the Realtek auto-parser.

   - The mute-LED control is standardized over all HD-audio codec
     drivers using the extended vmaster hook.

   - The vmaster slave mixer elements are initialized to 0dB as default
     so that the user won't be annoyed by the silent output after
     updates, e.g.  due to the additions of new elements.

   - Other many fix-ups for the misc HD-audio devices.

  In the ASoC side, this is a very active release, including a quite a
  few framework enhancements.  Some highlights:

   - Support for widgets not associated with a CODEC, an important part
     of the dynamic PCM framework.

   - A library factoring out the common code shared by dmaengine based
     DMA drivers contributed by Lars-Peter Clausen.  This will save a
     lot of code and make it much easier to deploy enhancements to
     dmaengine.

   - Support for binary controls, used for providing runtime
     configuration of algorithm coefficients.

   - A new DAPM widget type for regulator supplies allowing drivers for
     devices that can power down unused supplies while active to do
     without any per-driver code.

   - DAPM widgets for DAIs, initially giving a speed boost for playback
     startup and shutdown and also the basis for CODEC<->CODEC DAI link
     support.

   - Support for specifying the number of significant bits on audio
     interfaces, useful for allowing applications to know how much
     effort to put into generating data for a larger sample format.

   - Conversion of the FSI driver used on some SH processors to
     DMAEngine.

   - Conversion of EP93xx drivers to DMAEngine.

   - New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
     WM2200.

   - Move audmux driver from arc/arm to sound/soc

   - McBSP move from arch/ to sound/ and updates

  Also, a few small updates and fixes for other drivers like au88x0,
  ymfpci, USB 6fire, USB usx2yaudio are included."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits)
  ASoC: wm8994: Provide VMID mode control and fix default sequence
  ASoC: wm8994: Add missing break in resume
  ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
  ASoC: pxa-ssp: atomically set stream active masks
  ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
  ASoC: Samsung: Added to support mono recording
  ALSA: hda - Fix build with CONFIG_PM=n
  ALSA: au88x0 - Avoid possible Oops at unbinding
  ALSA: usb-audio - Fix build error by consitification of rate list
  ASoC: core: Fix obscure leak of runtime array
  ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link()
  ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
  ASoC: wm8996: Add 44.1kHz support
  ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE
  ASoC: mx27vis-aic32x4: Convert it to platform driver
  ALSA: hda - fix printing of high HDMI sample rates
  ALSA: ymfpci - Fix legacy registers on S3/S4 resume
  ALSA: control - Fixe a trailing white space error
  ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook()
  ALSA: hda - Add "Mute-LED Mode" enum control
  ...
This commit is contained in:
Linus Torvalds 2012-03-22 13:00:13 -07:00
parent 424a6f6ef9 6681bc0deb
commit b2094ef840
288 changed files with 17205 additions and 12532 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

24
Documentation/devicetree/bindings/sound/alc5632.txt Normal file
View File

@ -0,0 +1,24 @@
ALC5632 audio CODEC
This device supports I2C only.
Required properties:
- compatible : "realtek,alc5632"
- reg : the I2C address of the device.
- gpio-controller : Indicates this device is a GPIO controller.
- #gpio-cells : Should be two. The first cell is the pin number and the
second cell is used to specify optional parameters (currently unused).
Example:
alc5632: alc5632@1e {
compatible = "realtek,alc5632";
reg = <0x1a>;
gpio-controller;
#gpio-cells = <2>;
};

13
Documentation/devicetree/bindings/sound/imx-audmux.txt Normal file
View File

@ -0,0 +1,13 @@
Freescale Digital Audio Mux (AUDMUX) device
Required properties:
- compatible : "fsl,imx21-audmux" for AUDMUX version firstly used on i.MX21,
or "fsl,imx31-audmux" for the version firstly used on i.MX31.
- reg : Should contain AUDMUX registers location and length
Example:
audmux@021d8000 {
compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux";
reg = <0x021d8000 0x4000>;
};

0
Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt → Documentation/devicetree/bindings/sound/sgtl5000.txt
View File

59
Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt Normal file
View File

@ -0,0 +1,59 @@
NVIDIA Tegra audio complex
Required properties:
- compatible : "nvidia,tegra-audio-alc5632"
- nvidia,model : The user-visible name of this sound complex.
- nvidia,audio-routing : A list of the connections between audio components.
Each entry is a pair of strings, the first being the connection's sink,
the second being the connection's source. Valid names for sources and
sinks are the ALC5632's pins:
ALC5632 pins:
* SPK_OUTP
* SPK_OUTN
* HP_OUT_L
* HP_OUT_R
* AUX_OUT_P
* AUX_OUT_N
* LINE_IN_L
* LINE_IN_R
* PHONE_P
* PHONE_N
* MIC1_P
* MIC1_N
* MIC2_P
* MIC2_N
* MICBIAS1
* DMICDAT
Board connectors:
* Headset Stereophone
* Int Spk
* Headset Mic
* Digital Mic
- nvidia,i2s-controller : The phandle of the Tegra I2S controller
- nvidia,audio-codec : The phandle of the ALC5632 audio codec
Example:
sound {
compatible = "nvidia,tegra-audio-alc5632-paz00",
"nvidia,tegra-audio-alc5632";
nvidia,model = "Compal PAZ00";
nvidia,audio-routing =
"Int Spk", "SPK_OUTP",
"Int Spk", "SPK_OUTN",
"Headset Mic","MICBIAS1",
"MIC1_N", "Headset Mic",
"MIC1_P", "Headset Mic",
"Headset Stereophone", "HP_OUT_R",
"Headset Stereophone", "HP_OUT_L";
nvidia,i2s-controller = <&tegra_i2s1>;
nvidia,audio-codec = <&alc5632>;
};

1
Documentation/devicetree/bindings/vendor-prefixes.txt
View File

@ -34,6 +34,7 @@ picochip Picochip Ltd
powervr Imagination Technologies
qcom Qualcomm, Inc.
ramtron Ramtron International
realtek Realtek Semiconductor Corp.
samsung Samsung Semiconductor
sbs Smart Battery System
schindler Schindler

8
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@ -860,7 +860,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
[Multiple options for each card instance]
model - force the model name
position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF)
position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF,
3 = VIACOMBO, 4 = COMBO)
probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
When the bit 8 (0x100) is set, the lower 8 bits are used
as the "fixed" codec slots; i.e. the driver probes the
@ -925,6 +926,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
(Usually SD_LPIB register is more accurate than the
position buffer.)
position_fix=3 is specific to VIA devices. The position
of the capture stream is checked from both LPIB and POSBUF
values. position_fix=4 is a combination mode, using LPIB
for playback and POSBUF for capture.
NB: If you get many "azx_get_response timeout" messages at
loading, it's likely a problem of interrupts (e.g. ACPI irq
routing). Try to boot with options like "pci=noacpi". Also, you

79
Documentation/sound/alsa/HD-Audio-Models.txt
View File

@ -8,37 +8,10 @@ ALC880
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
======
fujitsu Fujitsu S7020
acer Acer TravelMate
will Will laptops (PB V7900)
replacer Replacer 672V
favorit100 Maxdata Favorit 100XS
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)
N/A
ALC262
======
@ -70,55 +43,7 @@ ALC680
ALC882/883/885/888/889
======================
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
mb5 Macbook 5,1
macmini3 Macmini 3,1
mba21 Macbook Air 2,1
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
imac91 iMac 9,1
w2jc ASUS W2JC
3stack-2ch-dig 3-jack with SPDIF I/O (ALC883)
alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883)
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
acer-aspire-6530g Acer Aspire 6530G
acer-aspire-7730g Acer Aspire 7730G
acer-aspire-8930g Acer Aspire 8930G
medion Medion Laptops
targa-dig Targa/MSI
targa-2ch-dig Targa/MSI with 2-channel
targa-8ch-dig Targa/MSI with 8-channel (MSI GX620)
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-m540r Clevo M540R (6ch + digital)
clevo-m720 Clevo M720 laptop series
fujitsu-pi2515 Fujitsu AMILO Pi2515
fujitsu-xa3530 Fujitsu AMILO XA3530
3stack-6ch-intel Intel DG33* boards
intel-alc889a Intel IbexPeak with ALC889A
intel-x58 Intel DX58 with ALC889
asus-p5q ASUS P5Q-EM boards
mb31 MacBook 3,1
sony-vaio-tt Sony VAIO TT
auto auto-config reading BIOS (default)
N/A
ALC861/660
==========

7
Documentation/sound/alsa/HD-Audio.txt
View File

@ -59,7 +59,12 @@ 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.
`position_fix=3` means to use a combination of both methods, needed
for some VIA and ATI controllers. 0 is the default value for all other
for some VIA controllers. The capture stream position is corrected
by comparing both LPIB and position-buffer values.
`position_fix=4` is another combination available for all controllers,
and uses LPIB for the playback and the position-buffer for the capture
streams.
0 is the default value for all other
controllers, the automatic check and fallback to LPIB as described in
the above. If you get a problem of repeated sounds, this option might
help.

19
arch/arm/mach-ep93xx/core.c
View File

@ -817,23 +817,12 @@ void __init ep93xx_register_i2s(void)
#define EP93XX_I2SCLKDIV_MASK (EP93XX_SYSCON_I2SCLKDIV_ORIDE | \
EP93XX_SYSCON_I2SCLKDIV_SPOL)
int ep93xx_i2s_acquire(unsigned i2s_pins, unsigned i2s_config)
int ep93xx_i2s_acquire(void)
{
unsigned val;
/* Sanity check */
if (i2s_pins & ~EP93XX_SYSCON_DEVCFG_I2S_MASK)
return -EINVAL;
if (i2s_config & ~EP93XX_I2SCLKDIV_MASK)
return -EINVAL;
/* Must have only one of I2SONSSP/I2SONAC97 set */
if ((i2s_pins & EP93XX_SYSCON_DEVCFG_I2SONSSP) ==
(i2s_pins & EP93XX_SYSCON_DEVCFG_I2SONAC97))
return -EINVAL;
ep93xx_devcfg_clear_bits(EP93XX_SYSCON_DEVCFG_I2S_MASK);
ep93xx_devcfg_set_bits(i2s_pins);
ep93xx_devcfg_set_clear(EP93XX_SYSCON_DEVCFG_I2SONAC97,
EP93XX_SYSCON_DEVCFG_I2S_MASK);
/*
* This is potentially racy with the clock api for i2s_mclk, sclk and
@ -843,7 +832,7 @@ int ep93xx_i2s_acquire(unsigned i2s_pins, unsigned i2s_config)
*/
val = __raw_readl(EP93XX_SYSCON_I2SCLKDIV);
val &= ~EP93XX_I2SCLKDIV_MASK;
val |= i2s_config;
val |= EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL;
ep93xx_syscon_swlocked_write(val, EP93XX_SYSCON_I2SCLKDIV);
return 0;

2
arch/arm/mach-ep93xx/include/mach/platform.h
View File

@ -59,7 +59,7 @@ void ep93xx_register_keypad(struct ep93xx_keypad_platform_data *data);
int ep93xx_keypad_acquire_gpio(struct platform_device *pdev);
void ep93xx_keypad_release_gpio(struct platform_device *pdev);
void ep93xx_register_i2s(void);
int ep93xx_i2s_acquire(unsigned i2s_pins, unsigned i2s_config);
int ep93xx_i2s_acquire(void);
void ep93xx_i2s_release(void);
void ep93xx_register_ac97(void);

6
arch/arm/mach-imx/Kconfig
View File

@ -46,7 +46,6 @@ config SOC_IMX21
bool
select MACH_MX21
select CPU_ARM926T
select ARCH_MXC_AUDMUX_V1
select IMX_HAVE_DMA_V1
select IMX_HAVE_IOMUX_V1
select MXC_AVIC
@ -55,7 +54,6 @@ config SOC_IMX25
bool
select ARCH_MX25
select CPU_ARM926T
select ARCH_MXC_AUDMUX_V2
select ARCH_MXC_IOMUX_V3
select MXC_AVIC
@ -63,7 +61,6 @@ config SOC_IMX27
bool
select MACH_MX27
select CPU_ARM926T
select ARCH_MXC_AUDMUX_V1
select IMX_HAVE_DMA_V1
select IMX_HAVE_IOMUX_V1
select MXC_AVIC
@ -72,7 +69,6 @@ config SOC_IMX31
bool
select CPU_V6
select IMX_HAVE_PLATFORM_MXC_RNGA
select ARCH_MXC_AUDMUX_V2
select MXC_AVIC
select SMP_ON_UP if SMP
@ -80,7 +76,6 @@ config SOC_IMX35
bool
select CPU_V6
select ARCH_MXC_IOMUX_V3
select ARCH_MXC_AUDMUX_V2
select HAVE_EPIT
select MXC_AVIC
select SMP_ON_UP if SMP
@ -89,7 +84,6 @@ config SOC_IMX5
select CPU_V7
select MXC_TZIC
select ARCH_MXC_IOMUX_V3
select ARCH_MXC_AUDMUX_V2
select ARCH_HAS_CPUFREQ
select ARCH_MX5
bool

20
arch/arm/mach-imx/eukrea_mbimx27-baseboard.c
View File

@ -32,7 +32,6 @@
#include <mach/common.h>
#include <mach/iomux-mx27.h>
#include <mach/hardware.h>
#include <mach/audmux.h>
#include "devices-imx27.h"
@ -306,25 +305,6 @@ void __init eukrea_mbimx27_baseboard_init(void)
mxc_gpio_setup_multiple_pins(eukrea_mbimx27_pins,
ARRAY_SIZE(eukrea_mbimx27_pins), "MBIMX27");
#if defined(CONFIG_SND_SOC_EUKREA_TLV320) \
|| defined(CONFIG_SND_SOC_EUKREA_TLV320_MODULE)
/* SSI unit master I2S codec connected to SSI_PINS_4*/
mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
MXC_AUDMUX_V1_PCR_SYN |
MXC_AUDMUX_V1_PCR_TFSDIR |
MXC_AUDMUX_V1_PCR_TCLKDIR |
MXC_AUDMUX_V1_PCR_RFSDIR |
MXC_AUDMUX_V1_PCR_RCLKDIR |
MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4)
);
mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4,
MXC_AUDMUX_V1_PCR_SYN |
MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
);
#endif
imx27_add_imx_uart1(&uart_pdata);
imx27_add_imx_uart2(&uart_pdata);
#if !defined(MACH_EUKREA_CPUIMX27_USEUART4)

1
arch/arm/mach-imx/eukrea_mbimxsd-baseboard.c
View File

@ -37,7 +37,6 @@
#include <mach/hardware.h>
#include <mach/common.h>
#include <mach/iomux-mx51.h>
#include <mach/audmux.h>
#include "devices-imx51.h"

17
arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c
View File

@ -31,7 +31,6 @@
#include <asm/mach-types.h>
#include <asm/mach/arch.h>
#include <mach/mx25.h>
#include <mach/audmux.h>
#include "devices-imx25.h"
@ -241,22 +240,6 @@ void __init eukrea_mbimxsd25_baseboard_init(void)
ARRAY_SIZE(eukrea_mbimxsd_pads)))
printk(KERN_ERR "error setting mbimxsd pads !\n");
#if defined(CONFIG_SND_SOC_EUKREA_TLV320)
/* SSI unit master I2S codec connected to SSI_AUD5*/
mxc_audmux_v2_configure_port(0,
MXC_AUDMUX_V2_PTCR_SYN |
MXC_AUDMUX_V2_PTCR_TFSDIR |
MXC_AUDMUX_V2_PTCR_TFSEL(4) |
MXC_AUDMUX_V2_PTCR_TCLKDIR |
MXC_AUDMUX_V2_PTCR_TCSEL(4),
MXC_AUDMUX_V2_PDCR_RXDSEL(4)
);
mxc_audmux_v2_configure_port(4,
MXC_AUDMUX_V2_PTCR_SYN,
MXC_AUDMUX_V2_PDCR_RXDSEL(0)
);
#endif
imx25_add_imx_uart1(&uart_pdata);
imx25_add_imx_fb(&eukrea_mximxsd_fb_pdata);
imx25_add_imx_ssi(0, &eukrea_mbimxsd_ssi_pdata);

17
arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c
View File

@ -38,7 +38,6 @@
#include <mach/hardware.h>
#include <mach/common.h>
#include <mach/iomux-mx35.h>
#include <mach/audmux.h>
#include "devices-imx35.h"
@ -252,22 +251,6 @@ void __init eukrea_mbimxsd35_baseboard_init(void)
ARRAY_SIZE(eukrea_mbimxsd_pads)))
printk(KERN_ERR "error setting mbimxsd pads !\n");
#if defined(CONFIG_SND_SOC_EUKREA_TLV320)
/* SSI unit master I2S codec connected to SSI_AUD4 */
mxc_audmux_v2_configure_port(0,
MXC_AUDMUX_V2_PTCR_SYN |
MXC_AUDMUX_V2_PTCR_TFSDIR |
MXC_AUDMUX_V2_PTCR_TFSEL(3) |
MXC_AUDMUX_V2_PTCR_TCLKDIR |
MXC_AUDMUX_V2_PTCR_TCSEL(3),
MXC_AUDMUX_V2_PDCR_RXDSEL(3)
);
mxc_audmux_v2_configure_port(3,
MXC_AUDMUX_V2_PTCR_SYN,
MXC_AUDMUX_V2_PDCR_RXDSEL(0)
);
#endif
imx35_add_imx_uart1(&uart_pdata);
imx35_add_ipu_core(&mx3_ipu_data);
imx35_add_mx3_sdc_fb(&mx3fb_pdata);

1
arch/arm/mach-imx/mach-imx27_visstrim_m10.c
View File

@ -263,6 +263,7 @@ static void __init visstrim_m10_board_init(void)
imx27_add_fec(NULL);
imx_add_gpio_keys(&visstrim_gpio_keys_platform_data);
platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices));
imx_add_platform_device("mx27vis", 0, NULL, 0, NULL, 0);
}
static void __init visstrim_m10_timer_init(void)

13
arch/arm/mach-imx/mach-pca100.c
View File

@ -36,7 +36,6 @@
#include <mach/hardware.h>
#include <mach/iomux-mx27.h>
#include <asm/mach/time.h>
#include <mach/audmux.h>
#include <mach/irqs.h>
#include <mach/ulpi.h>
@ -359,18 +358,6 @@ static void __init pca100_init(void)
imx27_soc_init();
/* SSI unit */
mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */
MXC_AUDMUX_V1_PCR_TFCSEL(3) |
MXC_AUDMUX_V1_PCR_TCLKDIR | /* clock is output */
MXC_AUDMUX_V1_PCR_RXDSEL(3));
mxc_audmux_v1_configure_port(3,
MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */
MXC_AUDMUX_V1_PCR_TFCSEL(0) |
MXC_AUDMUX_V1_PCR_TFSDIR |
MXC_AUDMUX_V1_PCR_RXDSEL(0));
ret = mxc_gpio_setup_multiple_pins(pca100_pins,
ARRAY_SIZE(pca100_pins), "PCA100");
if (ret)

13
arch/arm/mach-imx/mach-pcm043.c
View File

@ -37,7 +37,6 @@
#include <mach/common.h>
#include <mach/iomux-mx35.h>
#include <mach/ulpi.h>
#include <mach/audmux.h>
#include "devices-imx35.h"
@ -362,18 +361,6 @@ static void __init pcm043_init(void)
mxc_iomux_v3_setup_multiple_pads(pcm043_pads, ARRAY_SIZE(pcm043_pads));
mxc_audmux_v2_configure_port(3,
MXC_AUDMUX_V2_PTCR_SYN | /* 4wire mode */
MXC_AUDMUX_V2_PTCR_TFSEL(0) |
MXC_AUDMUX_V2_PTCR_TFSDIR,
MXC_AUDMUX_V2_PDCR_RXDSEL(0));
mxc_audmux_v2_configure_port(0,
MXC_AUDMUX_V2_PTCR_SYN | /* 4wire mode */
MXC_AUDMUX_V2_PTCR_TCSEL(3) |
MXC_AUDMUX_V2_PTCR_TCLKDIR, /* clock is output */
MXC_AUDMUX_V2_PDCR_RXDSEL(3));
imx35_add_fec(NULL);
platform_add_devices(devices, ARRAY_SIZE(devices));
imx35_add_imx2_wdt(NULL);

6
arch/arm/mach-imx/mm-imx21.c
View File

@ -75,6 +75,10 @@ void __init mx21_init_irq(void)
mxc_init_irq(MX21_IO_ADDRESS(MX21_AVIC_BASE_ADDR));
}
static const struct resource imx21_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX21_AUDMUX_BASE_ADDR, SZ_4K),
};
void __init imx21_soc_init(void)
{
mxc_register_gpio("imx21-gpio", 0, MX21_GPIO1_BASE_ADDR, SZ_256, MX21_INT_GPIO, 0);
@ -85,4 +89,6 @@ void __init imx21_soc_init(void)
mxc_register_gpio("imx21-gpio", 5, MX21_GPIO6_BASE_ADDR, SZ_256, MX21_INT_GPIO, 0);
imx_add_imx_dma();
platform_device_register_simple("imx21-audmux", 0, imx21_audmux_res,
ARRAY_SIZE(imx21_audmux_res));
}

7
arch/arm/mach-imx/mm-imx25.c
View File

@ -83,6 +83,10 @@ static struct sdma_platform_data imx25_sdma_pdata __initdata = {
.script_addrs = &imx25_sdma_script,
};
static const struct resource imx25_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX25_AUDMUX_BASE_ADDR, SZ_16K),
};
void __init imx25_soc_init(void)
{
/* i.mx25 has the i.mx31 type gpio */
@ -93,4 +97,7 @@ void __init imx25_soc_init(void)
/* i.mx25 has the i.mx35 type sdma */
imx_add_imx_sdma("imx35-sdma", MX25_SDMA_BASE_ADDR, MX25_INT_SDMA, &imx25_sdma_pdata);
/* i.mx25 has the i.mx31 type audmux */
platform_device_register_simple("imx31-audmux", 0, imx25_audmux_res,
ARRAY_SIZE(imx25_audmux_res));
}

7
arch/arm/mach-imx/mm-imx27.c
View File

@ -75,6 +75,10 @@ void __init mx27_init_irq(void)
mxc_init_irq(MX27_IO_ADDRESS(MX27_AVIC_BASE_ADDR));
}
static const struct resource imx27_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX27_AUDMUX_BASE_ADDR, SZ_4K),
};
void __init imx27_soc_init(void)
{
/* i.mx27 has the i.mx21 type gpio */
@ -86,4 +90,7 @@ void __init imx27_soc_init(void)
mxc_register_gpio("imx21-gpio", 5, MX27_GPIO6_BASE_ADDR, SZ_256, MX27_INT_GPIO, 0);
imx_add_imx_dma();
/* imx27 has the imx21 type audmux */
platform_device_register_simple("imx21-audmux", 0, imx27_audmux_res,
ARRAY_SIZE(imx27_audmux_res));
}

13
arch/arm/mach-imx/mm-imx3.c
View File

@ -158,6 +158,10 @@ static struct sdma_platform_data imx31_sdma_pdata __initdata = {
.script_addrs = &imx31_to2_sdma_script,
};
static const struct resource imx31_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX31_AUDMUX_BASE_ADDR, SZ_16K),
};
void __init imx31_soc_init(void)
{
int to_version = mx31_revision() >> 4;
@ -175,6 +179,8 @@ void __init imx31_soc_init(void)
}
imx_add_imx_sdma("imx31-sdma", MX31_SDMA_BASE_ADDR, MX31_INT_SDMA, &imx31_sdma_pdata);
platform_device_register_simple("imx31-audmux", 0, imx31_audmux_res,
ARRAY_SIZE(imx31_audmux_res));
}
#endif /* ifdef CONFIG_SOC_IMX31 */
@ -241,6 +247,10 @@ static struct sdma_platform_data imx35_sdma_pdata __initdata = {
.script_addrs = &imx35_to2_sdma_script,
};
static const struct resource imx35_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX35_AUDMUX_BASE_ADDR, SZ_16K),
};
void __init imx35_soc_init(void)
{
int to_version = mx35_revision() >> 4;
@ -259,5 +269,8 @@ void __init imx35_soc_init(void)
}
imx_add_imx_sdma("imx35-sdma", MX35_SDMA_BASE_ADDR, MX35_INT_SDMA, &imx35_sdma_pdata);
/* i.mx35 has the i.mx31 type audmux */
platform_device_register_simple("imx31-audmux", 0, imx35_audmux_res,
ARRAY_SIZE(imx35_audmux_res));
}
#endif /* ifdef CONFIG_SOC_IMX35 */

22
arch/arm/mach-imx/mm-imx5.c
View File

@ -170,6 +170,18 @@ static struct sdma_platform_data imx53_sdma_pdata __initdata = {
.script_addrs = &imx53_sdma_script,
};
static const struct resource imx50_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX50_AUDMUX_BASE_ADDR, SZ_16K),
};
static const struct resource imx51_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX51_AUDMUX_BASE_ADDR, SZ_16K),
};
static const struct resource imx53_audmux_res[] __initconst = {
DEFINE_RES_MEM(MX53_AUDMUX_BASE_ADDR, SZ_16K),
};
void __init imx50_soc_init(void)
{
/* i.mx50 has the i.mx31 type gpio */
@ -179,6 +191,10 @@ void __init imx50_soc_init(void)
mxc_register_gpio("imx31-gpio", 3, MX50_GPIO4_BASE_ADDR, SZ_16K, MX50_INT_GPIO4_LOW, MX50_INT_GPIO4_HIGH);
mxc_register_gpio("imx31-gpio", 4, MX50_GPIO5_BASE_ADDR, SZ_16K, MX50_INT_GPIO5_LOW, MX50_INT_GPIO5_HIGH);
mxc_register_gpio("imx31-gpio", 5, MX50_GPIO6_BASE_ADDR, SZ_16K, MX50_INT_GPIO6_LOW, MX50_INT_GPIO6_HIGH);
/* i.mx50 has the i.mx31 type audmux */
platform_device_register_simple("imx31-audmux", 0, imx50_audmux_res,
ARRAY_SIZE(imx50_audmux_res));
}
void __init imx51_soc_init(void)
@ -191,6 +207,9 @@ void __init imx51_soc_init(void)
/* i.mx51 has the i.mx35 type sdma */
imx_add_imx_sdma("imx35-sdma", MX51_SDMA_BASE_ADDR, MX51_INT_SDMA, &imx51_sdma_pdata);
/* i.mx51 has the i.mx31 type audmux */
platform_device_register_simple("imx31-audmux", 0, imx51_audmux_res,
ARRAY_SIZE(imx51_audmux_res));
}
void __init imx53_soc_init(void)
@ -206,4 +225,7 @@ void __init imx53_soc_init(void)
/* i.mx53 has the i.mx35 type sdma */
imx_add_imx_sdma("imx35-sdma", MX53_SDMA_BASE_ADDR, MX53_INT_SDMA, &imx53_sdma_pdata);
/* i.mx53 has the i.mx31 type audmux */
platform_device_register_simple("imx31-audmux", 0, imx53_audmux_res,
ARRAY_SIZE(imx53_audmux_res));
}

6
arch/arm/mach-kirkwood/openrd-setup.c
View File

@ -83,6 +83,11 @@ static struct i2c_board_info i2c_board_info[] __initdata = {
},
};
static struct platform_device openrd_client_audio_device = {
.name = "openrd-client-audio",
.id = -1,
};
static int __initdata uart1;
static int __init sd_uart_selection(char *str)
@ -172,6 +177,7 @@ static void __init openrd_init(void)
kirkwood_i2c_init();
if (machine_is_openrd_client() || machine_is_openrd_ultimate()) {
platform_device_register(&openrd_client_audio_device);
i2c_register_board_info(0, i2c_board_info,
ARRAY_SIZE(i2c_board_info));
kirkwood_audio_init();

6
arch/arm/mach-kirkwood/t5325-setup.c
View File

@ -106,6 +106,11 @@ static struct platform_device hp_t5325_button_device = {
}
};
static struct platform_device hp_t5325_audio_device = {
.name = "t5325-audio",
.id = -1,
};
static unsigned int hp_t5325_mpp_config[] __initdata = {
MPP0_NF_IO2,
MPP1_SPI_MOSI,
@ -179,6 +184,7 @@ static void __init hp_t5325_init(void)
kirkwood_sata_init(&hp_t5325_sata_data);
kirkwood_ehci_init();
platform_device_register(&hp_t5325_button_device);
platform_device_register(&hp_t5325_audio_device);
i2c_register_board_info(0, i2c_board_info, ARRAY_SIZE(i2c_board_info));
kirkwood_audio_init();

3
arch/arm/mach-omap1/Kconfig
View File

@ -37,7 +37,6 @@ comment "OMAP Board Type"
config MACH_OMAP_INNOVATOR
bool "TI Innovator"
depends on ARCH_OMAP1 && (ARCH_OMAP15XX || ARCH_OMAP16XX)
select OMAP_MCBSP
help
TI OMAP 1510 or 1610 Innovator board support. Say Y here if you
have such a board.
@ -45,7 +44,6 @@ config MACH_OMAP_INNOVATOR
config MACH_OMAP_H2
bool "TI H2 Support"
depends on ARCH_OMAP1 && ARCH_OMAP16XX
select OMAP_MCBSP
help
TI OMAP 1610/1611B H2 board support. Say Y here if you have such
a board.
@ -72,7 +70,6 @@ config MACH_HERALD
config MACH_OMAP_OSK
bool "TI OSK Support"
depends on ARCH_OMAP1 && ARCH_OMAP16XX
select OMAP_MCBSP
help
TI OMAP 5912 OSK (OMAP Starter Kit) board support. Say Y here
if you have such a board.

4
arch/arm/mach-omap1/Makefile
View File

@ -6,7 +6,9 @@
obj-y := io.o id.o sram.o time.o irq.o mux.o flash.o serial.o devices.o dma.o
obj-y += clock.o clock_data.o opp_data.o reset.o pm_bus.o timer.o
obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o
ifneq ($(CONFIG_SND_OMAP_SOC_MCBSP),)
obj-y += mcbsp.o
endif
obj-$(CONFIG_OMAP_32K_TIMER) += timer32k.o

9
arch/arm/mach-omap1/devices.c
View File

@ -28,7 +28,6 @@
#include <plat/mux.h>
#include <plat/mmc.h>
#include <plat/omap7xx.h>
#include <plat/mcbsp.h>
#include "clock.h"
@ -250,16 +249,8 @@ static struct platform_device omap_pcm = {
.id = -1,
};
OMAP_MCBSP_PLATFORM_DEVICE(1);
OMAP_MCBSP_PLATFORM_DEVICE(2);
OMAP_MCBSP_PLATFORM_DEVICE(3);
static void omap_init_audio(void)
{
platform_device_register(&omap_mcbsp1);
platform_device_register(&omap_mcbsp2);
if (!cpu_is_omap7xx())
platform_device_register(&omap_mcbsp3);
platform_device_register(&omap_pcm);
}

14
arch/arm/mach-omap1/mcbsp.c
View File

@ -419,18 +419,6 @@ static int __init omap1_mcbsp_init(void)
if (!cpu_class_is_omap1())
return -ENODEV;
if (cpu_is_omap7xx())
omap_mcbsp_count = OMAP7XX_MCBSP_COUNT;
else if (cpu_is_omap15xx())
omap_mcbsp_count = OMAP15XX_MCBSP_COUNT;
else if (cpu_is_omap16xx())
omap_mcbsp_count = OMAP16XX_MCBSP_COUNT;
mcbsp_ptr = kzalloc(omap_mcbsp_count * sizeof(struct omap_mcbsp *),
GFP_KERNEL);
if (!mcbsp_ptr)
return -ENOMEM;
if (cpu_is_omap7xx())
omap_mcbsp_register_board_cfg(omap7xx_mcbsp_res_0,
OMAP7XX_MCBSP_RES_SZ,
@ -449,7 +437,7 @@ static int __init omap1_mcbsp_init(void)
omap16xx_mcbsp_pdata,
OMAP16XX_MCBSP_COUNT);
return omap_mcbsp_init();
return 0;
}
arch_initcall(omap1_mcbsp_init);

4
arch/arm/mach-omap2/Makefile
View File

@ -17,7 +17,9 @@ obj-$(CONFIG_ARCH_OMAP2) += $(omap-2-3-common) $(hwmod-common)
obj-$(CONFIG_ARCH_OMAP3) += $(omap-2-3-common) $(hwmod-common) $(secure-common)
obj-$(CONFIG_ARCH_OMAP4) += prm44xx.o $(hwmod-common) $(secure-common)
obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o
ifneq ($(CONFIG_SND_OMAP_SOC_MCBSP),)
obj-y += mcbsp.o
endif
obj-$(CONFIG_TWL4030_CORE) += omap_twl.o

29
arch/arm/mach-omap2/board-4430sdp.c
View File

@ -41,6 +41,7 @@
#include <video/omap-panel-nokia-dsi.h>
#include <video/omap-panel-picodlp.h>
#include <linux/wl12xx.h>
#include <linux/platform_data/omap-abe-twl6040.h>
#include "mux.h"
#include "hsmmc.h"
@ -378,12 +379,40 @@ static struct platform_device sdp4430_dmic_codec = {
.id = -1,
};
static struct omap_abe_twl6040_data sdp4430_abe_audio_data = {
.card_name = "SDP4430",
.has_hs = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
.has_hf = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
.has_ep = 1,
.has_aux = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
.has_vibra = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
.has_dmic = 1,
.has_hsmic = 1,
.has_mainmic = 1,
.has_submic = 1,
.has_afm = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
.jack_detection = 1,
/* MCLK input is 38.4MHz */
.mclk_freq = 38400000,
};
static struct platform_device sdp4430_abe_audio = {
.name = "omap-abe-twl6040",
.id = -1,
.dev = {
.platform_data = &sdp4430_abe_audio_data,
},
};
static struct platform_device *sdp4430_devices[] __initdata = {
&sdp4430_gpio_keys_device,
&sdp4430_leds_gpio,
&sdp4430_leds_pwm,
&sdp4430_vbat,
&sdp4430_dmic_codec,
&sdp4430_abe_audio,
};
static struct omap_musb_board_data musb_board_data = {

60
arch/arm/mach-omap2/board-omap4panda.c
View File

@ -28,6 +28,7 @@
#include <linux/regulator/machine.h>
#include <linux/regulator/fixed.h>
#include <linux/wl12xx.h>
#include <linux/platform_data/omap-abe-twl6040.h>
#include <mach/hardware.h>
#include <asm/hardware/gic.h>
@ -91,9 +92,34 @@ static struct platform_device leds_gpio = {
},
};
static struct omap_abe_twl6040_data panda_abe_audio_data = {
/* Audio out */
.has_hs = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
/* HandsFree through expasion connector */
.has_hf = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
/* PandaBoard: FM TX, PandaBoardES: can be connected to audio out */
.has_aux = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
/* PandaBoard: FM RX, PandaBoardES: audio in */
.has_afm = ABE_TWL6040_LEFT | ABE_TWL6040_RIGHT,
/* No jack detection. */
.jack_detection = 0,
/* MCLK input is 38.4MHz */
.mclk_freq = 38400000,
};
static struct platform_device panda_abe_audio = {
.name = "omap-abe-twl6040",
.id = -1,
.dev = {
.platform_data = &panda_abe_audio_data,
},
};
static struct platform_device *panda_devices[] __initdata = {
&leds_gpio,
&wl1271_device,
&panda_abe_audio,
};
static const struct usbhs_omap_board_data usbhs_bdata __initconst = {
@ -252,8 +278,25 @@ static int __init omap4_twl6030_hsmmc_init(struct omap2_hsmmc_info *controllers)
return 0;
}
static struct twl4030_codec_data twl6040_codec = {
/* single-step ramp for headset and handsfree */
.hs_left_step = 0x0f,
.hs_right_step = 0x0f,
.hf_left_step = 0x1d,
.hf_right_step = 0x1d,
};
static struct twl4030_audio_data twl6040_audio = {
.codec = &twl6040_codec,
.audpwron_gpio = 127,
.naudint_irq = OMAP44XX_IRQ_SYS_2N,
.irq_base = TWL6040_CODEC_IRQ_BASE,
};
/* Panda board uses the common PMIC configuration */
static struct twl4030_platform_data omap4_panda_twldata;
static struct twl4030_platform_data omap4_panda_twldata = {
.audio = &twl6040_audio,
};
/*
* Display monitor features are burnt in their EEPROM as EDID data. The EEPROM
@ -485,6 +528,20 @@ void omap4_panda_display_init(void)
omap_mux_init_gpio(HDMI_GPIO_HPD, OMAP_PIN_INPUT_PULLDOWN);
}
static void omap4_panda_init_rev(void)
{
if (cpu_is_omap443x()) {
/* PandaBoard 4430 */
/* ASoC audio configuration */
panda_abe_audio_data.card_name = "PandaBoard";
panda_abe_audio_data.has_hsmic = 1;
} else {
/* PandaBoard ES */
/* ASoC audio configuration */
panda_abe_audio_data.card_name = "PandaBoardES";
}
}
static void __init omap4_panda_init(void)
{
int package = OMAP_PACKAGE_CBS;
@ -498,6 +555,7 @@ static void __init omap4_panda_init(void)
if (ret)
pr_err("error setting wl12xx data: %d\n", ret);
omap4_panda_init_rev();
omap4_panda_i2c_init();
platform_add_devices(panda_devices, ARRAY_SIZE(panda_devices));
platform_device_register(&omap_vwlan_device);

22
arch/arm/mach-omap2/devices.c
View File

@ -26,7 +26,6 @@
#include <plat/tc.h>
#include <plat/board.h>
#include <plat/mcbsp.h>
#include <plat/mmc.h>
#include <plat/dma.h>
#include <plat/omap_hwmod.h>
@ -304,29 +303,8 @@ static struct platform_device omap_pcm = {
.id = -1,
};
/*
* OMAP2420 has 2 McBSP ports
* OMAP2430 has 5 McBSP ports
* OMAP3 has 5 McBSP ports
* OMAP4 has 4 McBSP ports
*/
OMAP_MCBSP_PLATFORM_DEVICE(1);
OMAP_MCBSP_PLATFORM_DEVICE(2);
OMAP_MCBSP_PLATFORM_DEVICE(3);
OMAP_MCBSP_PLATFORM_DEVICE(4);
OMAP_MCBSP_PLATFORM_DEVICE(5);
static void omap_init_audio(void)
{
platform_device_register(&omap_mcbsp1);
platform_device_register(&omap_mcbsp2);
if (cpu_is_omap243x() || cpu_is_omap34xx() || cpu_is_omap44xx()) {
platform_device_register(&omap_mcbsp3);
platform_device_register(&omap_mcbsp4);
}
if (cpu_is_omap243x() || cpu_is_omap34xx())
platform_device_register(&omap_mcbsp5);
platform_device_register(&omap_pcm);
}

54
arch/arm/mach-omap2/mcbsp.c
View File

@ -34,7 +34,7 @@
#include "cm2xxx_3xxx.h"
#include "cm-regbits-34xx.h"
/* McBSP internal signal muxing function */
/* McBSP1 internal signal muxing function for OMAP2/3 */
static int omap2_mcbsp1_mux_rx_clk(struct device *dev, const char *signal,
const char *src)
{
@ -65,6 +65,42 @@ static int omap2_mcbsp1_mux_rx_clk(struct device *dev, const char *signal,
return 0;
}
/* McBSP4 internal signal muxing function for OMAP4 */
#define OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX (1 << 31)
#define OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX (1 << 30)
static int omap4_mcbsp4_mux_rx_clk(struct device *dev, const char *signal,
const char *src)
{
u32 v;
/*
* In CONTROL_MCBSPLP register only bit 30 (CLKR mux), and bit 31 (FSR
* mux) is used */
v = omap4_ctrl_pad_readl(OMAP4_CTRL_MODULE_PAD_CORE_CONTROL_MCBSPLP);
if (!strcmp(signal, "clkr")) {
if (!strcmp(src, "clkr"))
v &= ~OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX;
else if (!strcmp(src, "clkx"))
v |= OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX;
else
return -EINVAL;
} else if (!strcmp(signal, "fsr")) {
if (!strcmp(src, "fsr"))
v &= ~OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX;
else if (!strcmp(src, "fsx"))
v |= OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX;
else
return -EINVAL;
} else {
return -EINVAL;
}
omap4_ctrl_pad_writel(v, OMAP4_CTRL_MODULE_PAD_CORE_CONTROL_MCBSPLP);
return 0;
}
/* McBSP CLKS source switching function */
static int omap2_mcbsp_set_clk_src(struct device *dev, struct clk *clk,
const char *src)
@ -146,9 +182,15 @@ static int omap_init_mcbsp(struct omap_hwmod *oh, void *unused)
pdata->has_ccr = true;
}
pdata->set_clk_src = omap2_mcbsp_set_clk_src;
if (id == 1)
/* On OMAP2/3 the McBSP1 port has 6 pin configuration */
if (id == 1 && oh->class->rev < MCBSP_CONFIG_TYPE4)
pdata->mux_signal = omap2_mcbsp1_mux_rx_clk;
/* On OMAP4 the McBSP4 port has 6 pin configuration */
if (id == 4 && oh->class->rev == MCBSP_CONFIG_TYPE4)
pdata->mux_signal = omap4_mcbsp4_mux_rx_clk;
if (oh->class->rev == MCBSP_CONFIG_TYPE3) {
if (id == 2)
/* The FIFO has 1024 + 256 locations */
@ -180,7 +222,6 @@ static int omap_init_mcbsp(struct omap_hwmod *oh, void *unused)
name, oh->name);
return PTR_ERR(pdev);
}
omap_mcbsp_count++;
return 0;
}
@ -188,11 +229,6 @@ static int __init omap2_mcbsp_init(void)
{
omap_hwmod_for_each_by_class("mcbsp", omap_init_mcbsp, NULL);
mcbsp_ptr = kzalloc(omap_mcbsp_count * sizeof(struct omap_mcbsp *),
GFP_KERNEL);
if (!mcbsp_ptr)
return -ENOMEM;
return omap_mcbsp_init();
return 0;
}
arch_initcall(omap2_mcbsp_init);

1
arch/arm/mach-s3c64xx/mach-crag6410-module.c
View File

@ -102,6 +102,7 @@ static struct wm8962_pdata wm8962_pdata __initdata = {
0x8000 | WM8962_GPIO_FN_DMICDAT,
WM8962_GPIO_FN_IRQ, /* Open drain mode */
},
.in4_dc_measure = true,
};
static struct wm9081_pdata wm9081_pdata __initdata = {

30
arch/arm/mach-shmobile/board-ap4evb.c
View File

@ -737,26 +737,18 @@ fsi_set_rate_end:
return ret;
}
static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable)
{
int ret;
if (is_porta)
ret = fsi_ak4642_set_rate(dev, rate, enable);
else
ret = fsi_hdmi_set_rate(dev, rate, enable);
return ret;
}
static struct sh_fsi_platform_info fsi_info = {
.porta_flags = SH_FSI_BRS_INV,
.portb_flags = SH_FSI_BRS_INV |
SH_FSI_BRM_INV |
SH_FSI_LRS_INV |
SH_FSI_FMT_SPDIF,
.set_rate = fsi_set_rate,
.port_a = {
.flags = SH_FSI_BRS_INV,
.set_rate = fsi_ak4642_set_rate,
},
.port_b = {
.flags = SH_FSI_BRS_INV |
SH_FSI_BRM_INV |
SH_FSI_LRS_INV |
SH_FSI_FMT_SPDIF,
.set_rate = fsi_hdmi_set_rate,
},
};
static struct resource fsi_resources[] = {

18
arch/arm/mach-shmobile/board-mackerel.c
View File

@ -860,7 +860,7 @@ static int __fsi_set_round_rate(struct clk *clk, long rate, int enable)
return clk_enable(clk);
}
static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable)
static int fsi_b_set_rate(struct device *dev, int rate, int enable)
{
struct clk *fsib_clk;
struct clk *fdiv_clk = &sh7372_fsidivb_clk;
@ -869,10 +869,6 @@ static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable)
int ackmd_bpfmd;
int ret;
/* FSIA is slave mode. nothing to do here */
if (is_porta)
return 0;
/* clock start */
switch (rate) {
case 44100:
@ -916,14 +912,16 @@ fsi_set_rate_end:
}
static struct sh_fsi_platform_info fsi_info = {
.porta_flags = SH_FSI_BRS_INV,
.portb_flags = SH_FSI_BRS_INV |
.port_a = {
.flags = SH_FSI_BRS_INV,
},
.port_b = {
.flags = SH_FSI_BRS_INV |
SH_FSI_BRM_INV |
SH_FSI_LRS_INV |
SH_FSI_FMT_SPDIF,
.set_rate = fsi_set_rate,
.set_rate = fsi_b_set_rate,
}
};
static struct resource fsi_resources[] = {

6
arch/arm/plat-mxc/Kconfig
View File

@ -88,12 +88,6 @@ config IMX_HAVE_IOMUX_V1
config ARCH_MXC_IOMUX_V3
bool
config ARCH_MXC_AUDMUX_V1
bool
config ARCH_MXC_AUDMUX_V2
bool
config IRAM_ALLOC
bool
select GENERIC_ALLOCATOR

2
arch/arm/plat-mxc/Makefile
View File

@ -14,8 +14,6 @@ obj-$(CONFIG_IRAM_ALLOC) += iram_alloc.o
obj-$(CONFIG_MXC_PWM) += pwm.o
obj-$(CONFIG_MXC_ULPI) += ulpi.o
obj-$(CONFIG_MXC_USE_EPIT) += epit.o
obj-$(CONFIG_ARCH_MXC_AUDMUX_V1) += audmux-v1.o
obj-$(CONFIG_ARCH_MXC_AUDMUX_V2) += audmux-v2.o
obj-$(CONFIG_MXC_DEBUG_BOARD) += 3ds_debugboard.o
obj-$(CONFIG_CPU_FREQ_IMX) += cpufreq.o
ifdef CONFIG_SND_IMX_SOC

64
arch/arm/plat-mxc/audmux-v1.c
View File

@ -1,64 +0,0 @@
/*
* Copyright 2009 Pengutronix, Sascha Hauer <s.hauer@pengutronix.de>
*
* Initial development of this code was funded by
* Phytec Messtechnik GmbH, http://www.phytec.de
*
* 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.
*/
#include <linux/module.h>
#include <linux/err.h>
#include <linux/io.h>
#include <linux/clk.h>
#include <mach/audmux.h>
#include <mach/hardware.h>
static void __iomem *audmux_base;
static unsigned char port_mapping[] = {
0x0, 0x4, 0x8, 0x10, 0x14, 0x1c,
};
int mxc_audmux_v1_configure_port(unsigned int port, unsigned int pcr)
{
if (!audmux_base) {
printk("%s: not configured\n", __func__);
return -ENOSYS;
}
if (port >= ARRAY_SIZE(port_mapping))
return -EINVAL;
writel(pcr, audmux_base + port_mapping[port]);
return 0;
}
EXPORT_SYMBOL_GPL(mxc_audmux_v1_configure_port);
static int mxc_audmux_v1_init(void)
{
#ifdef CONFIG_MACH_MX21
if (cpu_is_mx21())
audmux_base = MX21_IO_ADDRESS(MX21_AUDMUX_BASE_ADDR);
else
#endif
#ifdef CONFIG_MACH_MX27
if (cpu_is_mx27())
audmux_base = MX27_IO_ADDRESS(MX27_AUDMUX_BASE_ADDR);
else
#endif
(void)0;
return 0;
}
postcore_initcall(mxc_audmux_v1_init);

60
arch/arm/plat-mxc/include/mach/audmux.h
View File

@ -1,60 +0,0 @@
#ifndef __MACH_AUDMUX_H
#define __MACH_AUDMUX_H
#define MX27_AUDMUX_HPCR1_SSI0 0
#define MX27_AUDMUX_HPCR2_SSI1 1
#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2
#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3
#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4
#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5
#define MX31_AUDMUX_PORT1_SSI0 0
#define MX31_AUDMUX_PORT2_SSI1 1
#define MX31_AUDMUX_PORT3_SSI_PINS_3 2
#define MX31_AUDMUX_PORT4_SSI_PINS_4 3
#define MX31_AUDMUX_PORT5_SSI_PINS_5 4
#define MX31_AUDMUX_PORT6_SSI_PINS_6 5
#define MX51_AUDMUX_PORT1_SSI0 0
#define MX51_AUDMUX_PORT2_SSI1 1
#define MX51_AUDMUX_PORT3 2
#define MX51_AUDMUX_PORT4 3
#define MX51_AUDMUX_PORT5 4
#define MX51_AUDMUX_PORT6 5
#define MX51_AUDMUX_PORT7 6
/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */
#define MXC_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff)
#define MXC_AUDMUX_V1_PCR_INMEN (1 << 8)
#define MXC_AUDMUX_V1_PCR_TXRXEN (1 << 10)
#define MXC_AUDMUX_V1_PCR_SYN (1 << 12)
#define MXC_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13)
#define MXC_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20)
#define MXC_AUDMUX_V1_PCR_RCLKDIR (1 << 24)
#define MXC_AUDMUX_V1_PCR_RFSDIR (1 << 25)
#define MXC_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26)
#define MXC_AUDMUX_V1_PCR_TCLKDIR (1 << 30)
#define MXC_AUDMUX_V1_PCR_TFSDIR (1 << 31)
/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */
#define MXC_AUDMUX_V2_PTCR_TFSDIR (1 << 31)
#define MXC_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27)
#define MXC_AUDMUX_V2_PTCR_TCLKDIR (1 << 26)
#define MXC_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22)
#define MXC_AUDMUX_V2_PTCR_RFSDIR (1 << 21)
#define MXC_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17)
#define MXC_AUDMUX_V2_PTCR_RCLKDIR (1 << 16)
#define MXC_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12)
#define MXC_AUDMUX_V2_PTCR_SYN (1 << 11)
#define MXC_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13)
#define MXC_AUDMUX_V2_PDCR_TXRXEN (1 << 12)
#define MXC_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8)
#define MXC_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff)
int mxc_audmux_v1_configure_port(unsigned int port, unsigned int pcr);
int mxc_audmux_v2_configure_port(unsigned int port, unsigned int ptcr,
unsigned int pdcr);
#endif /* __MACH_AUDMUX_H */

8
arch/arm/plat-omap/Kconfig
View File

@ -110,14 +110,6 @@ config OMAP_MUX_WARNINGS
to change the pin multiplexing setup. When there are no warnings
printed, it's safe to deselect OMAP_MUX for your product.
config OMAP_MCBSP
bool "McBSP support"
depends on ARCH_OMAP
default y
help
Say Y here if you want support for the OMAP Multichannel
Buffered Serial Port.
config OMAP_MBOX_FWK
tristate "Mailbox framework support"
depends on ARCH_OMAP

2
arch/arm/plat-omap/Makefile
View File

@ -17,8 +17,6 @@ obj-$(CONFIG_ARCH_OMAP2) += omap_device.o
obj-$(CONFIG_ARCH_OMAP3) += omap_device.o
obj-$(CONFIG_ARCH_OMAP4) += omap_device.o
obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o
obj-$(CONFIG_OMAP_DM_TIMER) += dmtimer.o
obj-$(CONFIG_OMAP_DEBUG_DEVICES) += debug-devices.o
obj-$(CONFIG_OMAP_DEBUG_LEDS) += debug-leds.o

333
arch/arm/plat-omap/include/plat/mcbsp.h
View File

@ -27,271 +27,10 @@
#include <linux/spinlock.h>
#include <linux/clk.h>
/* macro for building platform_device for McBSP ports */
#define OMAP_MCBSP_PLATFORM_DEVICE(port_nr) \
static struct platform_device omap_mcbsp##port_nr = { \
.name = "omap-mcbsp-dai", \
.id = port_nr - 1, \
}
#define MCBSP_CONFIG_TYPE2 0x2
#define MCBSP_CONFIG_TYPE3 0x3
#define MCBSP_CONFIG_TYPE4 0x4
/* McBSP register numbers. Register address offset = num * reg_step */
enum {
/* Common registers */
OMAP_MCBSP_REG_SPCR2 = 4,
OMAP_MCBSP_REG_SPCR1,
OMAP_MCBSP_REG_RCR2,
OMAP_MCBSP_REG_RCR1,
OMAP_MCBSP_REG_XCR2,
OMAP_MCBSP_REG_XCR1,
OMAP_MCBSP_REG_SRGR2,
OMAP_MCBSP_REG_SRGR1,
OMAP_MCBSP_REG_MCR2,
OMAP_MCBSP_REG_MCR1,
OMAP_MCBSP_REG_RCERA,
OMAP_MCBSP_REG_RCERB,
OMAP_MCBSP_REG_XCERA,
OMAP_MCBSP_REG_XCERB,
OMAP_MCBSP_REG_PCR0,
OMAP_MCBSP_REG_RCERC,
OMAP_MCBSP_REG_RCERD,
OMAP_MCBSP_REG_XCERC,
OMAP_MCBSP_REG_XCERD,
OMAP_MCBSP_REG_RCERE,
OMAP_MCBSP_REG_RCERF,
OMAP_MCBSP_REG_XCERE,
OMAP_MCBSP_REG_XCERF,
OMAP_MCBSP_REG_RCERG,
OMAP_MCBSP_REG_RCERH,
OMAP_MCBSP_REG_XCERG,
OMAP_MCBSP_REG_XCERH,
/* OMAP1-OMAP2420 registers */
OMAP_MCBSP_REG_DRR2 = 0,
OMAP_MCBSP_REG_DRR1,
OMAP_MCBSP_REG_DXR2,
OMAP_MCBSP_REG_DXR1,
/* OMAP2430 and onwards */
OMAP_MCBSP_REG_DRR = 0,
OMAP_MCBSP_REG_DXR = 2,
OMAP_MCBSP_REG_SYSCON = 35,
OMAP_MCBSP_REG_THRSH2,
OMAP_MCBSP_REG_THRSH1,
OMAP_MCBSP_REG_IRQST = 40,
OMAP_MCBSP_REG_IRQEN,
OMAP_MCBSP_REG_WAKEUPEN,
OMAP_MCBSP_REG_XCCR,
OMAP_MCBSP_REG_RCCR,
OMAP_MCBSP_REG_XBUFFSTAT,
OMAP_MCBSP_REG_RBUFFSTAT,
OMAP_MCBSP_REG_SSELCR,
};
/* OMAP3 sidetone control registers */
#define OMAP_ST_REG_REV 0x00
#define OMAP_ST_REG_SYSCONFIG 0x10
#define OMAP_ST_REG_IRQSTATUS 0x18
#define OMAP_ST_REG_IRQENABLE 0x1C
#define OMAP_ST_REG_SGAINCR 0x24
#define OMAP_ST_REG_SFIRCR 0x28
#define OMAP_ST_REG_SSELCR 0x2C
/************************** McBSP SPCR1 bit definitions ***********************/
#define RRST 0x0001
#define RRDY 0x0002
#define RFULL 0x0004
#define RSYNC_ERR 0x0008
#define RINTM(value) ((value)<<4) /* bits 4:5 */
#define ABIS 0x0040
#define DXENA 0x0080
#define CLKSTP(value) ((value)<<11) /* bits 11:12 */
#define RJUST(value) ((value)<<13) /* bits 13:14 */
#define ALB 0x8000
#define DLB 0x8000
/************************** McBSP SPCR2 bit definitions ***********************/
#define XRST 0x0001
#define XRDY 0x0002
#define XEMPTY 0x0004
#define XSYNC_ERR 0x0008
#define XINTM(value) ((value)<<4) /* bits 4:5 */
#define GRST 0x0040
#define FRST 0x0080
#define SOFT 0x0100
#define FREE 0x0200
/************************** McBSP PCR bit definitions *************************/
#define CLKRP 0x0001
#define CLKXP 0x0002
#define FSRP 0x0004
#define FSXP 0x0008
#define DR_STAT 0x0010
#define DX_STAT 0x0020
#define CLKS_STAT 0x0040
#define SCLKME 0x0080
#define CLKRM 0x0100
#define CLKXM 0x0200
#define FSRM 0x0400
#define FSXM 0x0800
#define RIOEN 0x1000
#define XIOEN 0x2000
#define IDLE_EN 0x4000
/************************** McBSP RCR1 bit definitions ************************/
#define RWDLEN1(value) ((value)<<5) /* Bits 5:7 */
#define RFRLEN1(value) ((value)<<8) /* Bits 8:14 */
/************************** McBSP XCR1 bit definitions ************************/
#define XWDLEN1(value) ((value)<<5) /* Bits 5:7 */
#define XFRLEN1(value) ((value)<<8) /* Bits 8:14 */
/*************************** McBSP RCR2 bit definitions ***********************/
#define RDATDLY(value) (value) /* Bits 0:1 */
#define RFIG 0x0004
#define RCOMPAND(value) ((value)<<3) /* Bits 3:4 */
#define RWDLEN2(value) ((value)<<5) /* Bits 5:7 */
#define RFRLEN2(value) ((value)<<8) /* Bits 8:14 */
#define RPHASE 0x8000
/*************************** McBSP XCR2 bit definitions ***********************/
#define XDATDLY(value) (value) /* Bits 0:1 */
#define XFIG 0x0004
#define XCOMPAND(value) ((value)<<3) /* Bits 3:4 */
#define XWDLEN2(value) ((value)<<5) /* Bits 5:7 */
#define XFRLEN2(value) ((value)<<8) /* Bits 8:14 */
#define XPHASE 0x8000
/************************* McBSP SRGR1 bit definitions ************************/
#define CLKGDV(value) (value) /* Bits 0:7 */
#define FWID(value) ((value)<<8) /* Bits 8:15 */
/************************* McBSP SRGR2 bit definitions ************************/
#define FPER(value) (value) /* Bits 0:11 */
#define FSGM 0x1000
#define CLKSM 0x2000
#define CLKSP 0x4000
#define GSYNC 0x8000
/************************* McBSP MCR1 bit definitions *************************/
#define RMCM 0x0001
#define RCBLK(value) ((value)<<2) /* Bits 2:4 */
#define RPABLK(value) ((value)<<5) /* Bits 5:6 */
#define RPBBLK(value) ((value)<<7) /* Bits 7:8 */
/************************* McBSP MCR2 bit definitions *************************/
#define XMCM(value) (value) /* Bits 0:1 */
#define XCBLK(value) ((value)<<2) /* Bits 2:4 */
#define XPABLK(value) ((value)<<5) /* Bits 5:6 */
#define XPBBLK(value) ((value)<<7) /* Bits 7:8 */
/*********************** McBSP XCCR bit definitions *************************/
#define EXTCLKGATE 0x8000
#define PPCONNECT 0x4000
#define DXENDLY(value) ((value)<<12) /* Bits 12:13 */
#define XFULL_CYCLE 0x0800
#define DILB 0x0020
#define XDMAEN 0x0008
#define XDISABLE 0x0001
/********************** McBSP RCCR bit definitions *************************/
#define RFULL_CYCLE 0x0800
#define RDMAEN 0x0008
#define RDISABLE 0x0001
/********************** McBSP SYSCONFIG bit definitions ********************/
#define CLOCKACTIVITY(value) ((value)<<8)
#define SIDLEMODE(value) ((value)<<3)
#define ENAWAKEUP 0x0004
#define SOFTRST 0x0002
/********************** McBSP SSELCR bit definitions ***********************/
#define SIDETONEEN 0x0400
/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/
#define ST_AUTOIDLE 0x0001
/********************** McBSP Sidetone SGAINCR bit definitions *************/
#define ST_CH1GAIN(value) ((value<<16)) /* Bits 16:31 */
#define ST_CH0GAIN(value) (value) /* Bits 0:15 */
/********************** McBSP Sidetone SFIRCR bit definitions **************/
#define ST_FIRCOEFF(value) (value) /* Bits 0:15 */
/********************** McBSP Sidetone SSELCR bit definitions **************/
#define ST_COEFFWRDONE 0x0004
#define ST_COEFFWREN 0x0002
#define ST_SIDETONEEN 0x0001
/********************** McBSP DMA operating modes **************************/
#define MCBSP_DMA_MODE_ELEMENT 0
#define MCBSP_DMA_MODE_THRESHOLD 1
#define MCBSP_DMA_MODE_FRAME 2
/********************** McBSP WAKEUPEN bit definitions *********************/
#define XEMPTYEOFEN 0x4000
#define XRDYEN 0x0400
#define XEOFEN 0x0200
#define XFSXEN 0x0100
#define XSYNCERREN 0x0080
#define RRDYEN 0x0008
#define REOFEN 0x0004
#define RFSREN 0x0002
#define RSYNCERREN 0x0001
/* CLKR signal muxing options */
#define CLKR_SRC_CLKR 0
#define CLKR_SRC_CLKX 1
/* FSR signal muxing options */
#define FSR_SRC_FSR 0
#define FSR_SRC_FSX 1
/* McBSP functional clock sources */
#define MCBSP_CLKS_PRCM_SRC 0
#define MCBSP_CLKS_PAD_SRC 1
/* we don't do multichannel for now */
struct omap_mcbsp_reg_cfg {
u16 spcr2;
u16 spcr1;
u16 rcr2;
u16 rcr1;
u16 xcr2;
u16 xcr1;
u16 srgr2;
u16 srgr1;
u16 mcr2;
u16 mcr1;
u16 pcr0;
u16 rcerc;
u16 rcerd;
u16 xcerc;
u16 xcerd;
u16 rcere;
u16 rcerf;
u16 xcere;
u16 xcerf;
u16 rcerg;
u16 rcerh;
u16 xcerg;
u16 xcerh;
u16 xccr;
u16 rccr;
};
typedef enum {
OMAP_MCBSP_WORD_8 = 0,
OMAP_MCBSP_WORD_12,
OMAP_MCBSP_WORD_16,
OMAP_MCBSP_WORD_20,
OMAP_MCBSP_WORD_24,
OMAP_MCBSP_WORD_32,
} omap_mcbsp_word_length;
/* Platform specific configuration */
struct omap_mcbsp_ops {
void (*request)(unsigned int);
@ -312,43 +51,6 @@ struct omap_mcbsp_platform_data {
int (*mux_signal)(struct device *dev, const char *signal, const char *src);
};
struct omap_mcbsp_st_data {
void __iomem *io_base_st;
bool running;
bool enabled;
s16 taps[128]; /* Sidetone filter coefficients */
int nr_taps; /* Number of filter coefficients in use */
s16 ch0gain;
s16 ch1gain;
};
struct omap_mcbsp {
struct device *dev;
unsigned long phys_base;
unsigned long phys_dma_base;
void __iomem *io_base;
u8 id;
u8 free;
int rx_irq;
int tx_irq;
/* DMA stuff */
u8 dma_rx_sync;
u8 dma_tx_sync;
/* Protect the field .free, while checking if the mcbsp is in use */
spinlock_t lock;
struct omap_mcbsp_platform_data *pdata;
struct clk *fclk;
struct omap_mcbsp_st_data *st_data;
int dma_op_mode;
u16 max_tx_thres;
u16 max_rx_thres;
void *reg_cache;
int reg_cache_size;
};
/**
* omap_mcbsp_dev_attr - OMAP McBSP device attributes for omap_hwmod
* @sidetone: name of the sidetone device
@ -357,39 +59,4 @@ struct omap_mcbsp_dev_attr {
const char *sidetone;
};
extern struct omap_mcbsp **mcbsp_ptr;
extern int omap_mcbsp_count;
int omap_mcbsp_init(void);
void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config);
void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold);
void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold);
u16 omap_mcbsp_get_max_tx_threshold(unsigned int id);
u16 omap_mcbsp_get_max_rx_threshold(unsigned int id);
u16 omap_mcbsp_get_fifo_size(unsigned int id);
u16 omap_mcbsp_get_tx_delay(unsigned int id);
u16 omap_mcbsp_get_rx_delay(unsigned int id);
int omap_mcbsp_get_dma_op_mode(unsigned int id);
int omap_mcbsp_request(unsigned int id);
void omap_mcbsp_free(unsigned int id);
void omap_mcbsp_start(unsigned int id, int tx, int rx);
void omap_mcbsp_stop(unsigned int id, int tx, int rx);
/* McBSP functional clock source changing function */
extern int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id);
/* McBSP signal muxing API */
void omap2_mcbsp1_mux_clkr_src(u8 mux);
void omap2_mcbsp1_mux_fsr_src(u8 mux);
int omap_mcbsp_dma_ch_params(unsigned int id, unsigned int stream);
int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream);
/* Sidetone specific API */
int omap_st_set_chgain(unsigned int id, int channel, s16 chgain);
int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain);
int omap_st_enable(unsigned int id);
int omap_st_disable(unsigned int id);
int omap_st_is_enabled(unsigned int id);
#endif

6
arch/mips/jz4740/board-qi_lb60.c
View File

@ -418,6 +418,11 @@ static struct platform_device qi_lb60_charger_device = {
},
};
/* audio */
static struct platform_device qi_lb60_audio_device = {
.name = "qi-lb60-audio",
.id = -1,
};
static struct platform_device *jz_platform_devices[] __initdata = {
&jz4740_udc_device,
@ -434,6 +439,7 @@ static struct platform_device *jz_platform_devices[] __initdata = {
&qi_lb60_gpio_keys,
&qi_lb60_pwm_beeper,
&qi_lb60_charger_device,
&qi_lb60_audio_device,
};
static void __init board_gpio_setup(void)

4
arch/sh/boards/mach-ecovec24/setup.c
View File

@ -769,7 +769,9 @@ static struct platform_device camera_devices[] = {
/* FSI */
static struct sh_fsi_platform_info fsi_info = {
.portb_flags = SH_FSI_BRS_INV,
.port_b = {
.flags = SH_FSI_BRS_INV,
},
};
static struct resource fsi_resources[] = {

4
arch/sh/boards/mach-se/7724/setup.c
View File

@ -278,7 +278,9 @@ static struct platform_device ceu1_device = {
/* FSI */
/* change J20, J21, J22 pin to 1-2 connection to use slave mode */
static struct sh_fsi_platform_info fsi_info = {
.porta_flags = SH_FSI_BRS_INV,
.port_a = {
.flags = SH_FSI_BRS_INV,
},
};
static struct resource fsi_resources[] = {

3
drivers/base/regmap/internal.h
View File

@ -75,6 +75,9 @@ struct regmap {
const void *reg_defaults_raw;
void *cache;
bool cache_dirty;
struct reg_default *patch;
int patch_regs;
};
struct regcache_ops {

11
drivers/base/regmap/regcache.c
View File

@ -268,6 +268,17 @@ int regcache_sync(struct regmap *map)
map->cache_ops->name);
name = map->cache_ops->name;
trace_regcache_sync(map->dev, name, "start");
/* Apply any patch first */
for (i = 0; i < map->patch_regs; i++) {
ret = _regmap_write(map, map->patch[i].reg, map->patch[i].def);
if (ret != 0) {
dev_err(map->dev, "Failed to write %x = %x: %d\n",
map->patch[i].reg, map->patch[i].def, ret);
goto out;
}
}
if (!map->cache_dirty)
goto out;
if (map->cache_ops->sync) {

73
drivers/base/regmap/regmap.c
View File

@ -672,6 +672,79 @@ int regmap_update_bits_check(struct regmap *map, unsigned int reg,
}
EXPORT_SYMBOL_GPL(regmap_update_bits_check);
/**
* regmap_register_patch: Register and apply register updates to be applied
* on device initialistion
*
* @map: Register map to apply updates to.
* @regs: Values to update.
* @num_regs: Number of entries in regs.
*
* Register a set of register updates to be applied to the device
* whenever the device registers are synchronised with the cache and
* apply them immediately. Typically this is used to apply
* corrections to be applied to the device defaults on startup, such
* as the updates some vendors provide to undocumented registers.
*/
int regmap_register_patch(struct regmap *map, const struct reg_default *regs,
int num_regs)
{
int i, ret;
bool bypass;
/* If needed the implementation can be extended to support this */
if (map->patch)
return -EBUSY;
mutex_lock(&map->lock);
bypass = map->cache_bypass;
map->cache_bypass = true;
/* Write out first; it's useful to apply even if we fail later. */
for (i = 0; i < num_regs; i++) {
ret = _regmap_write(map, regs[i].reg, regs[i].def);
if (ret != 0) {
dev_err(map->dev, "Failed to write %x = %x: %d\n",
regs[i].reg, regs[i].def, ret);
goto out;
}
}
map->patch = kcalloc(sizeof(struct reg_default), num_regs, GFP_KERNEL);
if (map->patch != NULL) {
memcpy(map->patch, regs,
num_regs * sizeof(struct reg_default));
map->patch_regs = num_regs;
} else {
ret = -ENOMEM;
}
out:
map->cache_bypass = bypass;
mutex_unlock(&map->lock);
return ret;
}
EXPORT_SYMBOL_GPL(regmap_register_patch);
/*
* regmap_get_val_bytes(): Report the size of a register value
*
* Report the size of a register value, mainly intended to for use by
* generic infrastructure built on top of regmap.
*/
int regmap_get_val_bytes(struct regmap *map)
{
if (map->format.format_write)
return -EINVAL;
return map->format.val_bytes;
}
EXPORT_SYMBOL_GPL(regmap_get_val_bytes);
static int __init regmap_initcall(void)
{
regmap_debugfs_initcall();

3
include/linux/mfd/wm8994/pdata.h
View File

@ -185,6 +185,9 @@ struct wm8994_pdata {
unsigned int jd_scthr:2;
unsigned int jd_thr:2;
/* Configure WM1811 jack detection for use with external capacitor */
unsigned int jd_ext_cap:1;
/* WM8958 microphone bias configuration */
int micbias[2];

49
include/linux/platform_data/omap-abe-twl6040.h Normal file
View File

@ -0,0 +1,49 @@
/**
* omap-abe-twl6040.h - ASoC machine driver OMAP4+ devices, header.
*
* Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com
* All rights reserved.
*
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* 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.
*
* 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., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef _OMAP_ABE_TWL6040_H_
#define _OMAP_ABE_TWL6040_H_
/* To select if only one channel is connected in a stereo port */
#define ABE_TWL6040_LEFT (1 << 0)
#define ABE_TWL6040_RIGHT (1 << 1)
struct omap_abe_twl6040_data {
char *card_name;
/* Feature flags for connected audio pins */
u8 has_hs;
u8 has_hf;
bool has_ep;
u8 has_aux;
u8 has_vibra;
bool has_dmic;
bool has_hsmic;
bool has_mainmic;
bool has_submic;
u8 has_afm;
/* Other features */
bool jack_detection; /* board can detect jack events */
int mclk_freq; /* MCLK frequency speed for twl6040 */
};
#endif /* _OMAP_ABE_TWL6040_H_ */

4
include/linux/regmap.h
View File

@ -143,12 +143,16 @@ int regmap_update_bits(struct regmap *map, unsigned int reg,
int regmap_update_bits_check(struct regmap *map, unsigned int reg,
unsigned int mask, unsigned int val,
bool *change);
int regmap_get_val_bytes(struct regmap *map);
int regcache_sync(struct regmap *map);
void regcache_cache_only(struct regmap *map, bool enable);
void regcache_cache_bypass(struct regmap *map, bool enable);
void regcache_mark_dirty(struct regmap *map);
int regmap_register_patch(struct regmap *map, const struct reg_default *regs,
int num_regs);
/**
* Description of an IRQ for the generic regmap irq_chip.
*

7
include/sound/control.h
View File

@ -40,7 +40,7 @@ struct snd_kcontrol_new {
snd_ctl_elem_iface_t iface; /* interface identifier */
unsigned int device; /* device/client number */
unsigned int subdevice; /* subdevice (substream) number */
unsigned char *name; /* ASCII name of item */
const unsigned char *name; /* ASCII name of item */
unsigned int index; /* index of item */
unsigned int access; /* access rights */
unsigned int count; /* count of same elements */
@ -227,6 +227,11 @@ snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
}
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
void (*hook)(void *private_data, int),
void *private_data);
void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kctl);
/*
* Helper functions for jack-detection controls
*/

49
include/sound/dmaengine_pcm.h Normal file
View File

@ -0,0 +1,49 @@
/*
* Copyright (C) 2012, Analog Devices Inc.
* Author: Lars-Peter Clausen <lars@metafoo.de>
*
* 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.
*
* 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.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#ifndef __SOUND_DMAENGINE_PCM_H__
#define __SOUND_DMAENGINE_PCM_H__
#include <sound/pcm.h>
#include <linux/dmaengine.h>
/**
* snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM
* substream
* @substream: PCM substream
*/
static inline enum dma_transfer_direction
snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream)
{
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return DMA_MEM_TO_DEV;
else
return DMA_DEV_TO_MEM;
}
void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data);
void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream);
int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config);
int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream);
int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
dma_filter_fn filter_fn, void *filter_data);
int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream);
struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream);
#endif

3
include/sound/jack.h
View File

@ -53,6 +53,9 @@ enum snd_jack_types {
SND_JACK_BTN_5 = 0x0200,
};
/* Keep in sync with definitions above */
#define SND_JACK_SWITCH_TYPES 6
struct snd_jack {
struct input_dev *input_dev;
int registered;

24
include/sound/max9768.h Normal file
View File

@ -0,0 +1,24 @@
/*
* Platform data for MAX9768
* Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K.
* same licence as the driver
*/
#ifndef __SOUND_MAX9768_PDATA_H__
#define __SOUND_MAX9768_PDATA_H__
/**
* struct max9768_pdata - optional platform specific MAX9768 configuration
* @shdn_gpio: GPIO to SHDN pin. If not valid, pin must be hardwired HIGH
* @mute_gpio: GPIO to MUTE pin. If not valid, control for mute won't be added
* @flags: configuration flags, e.g. set classic PWM mode (check datasheet
* regarding "filterless modulation" which is default).
*/
struct max9768_pdata {
int shdn_gpio;
int mute_gpio;
unsigned flags;
#define MAX9768_FLAG_CLASSIC_PWM (1 << 0)
};
#endif /* __SOUND_MAX9768_PDATA_H__*/

9
include/sound/pcm.h
View File

@ -264,7 +264,7 @@ struct snd_pcm_hw_constraint_ratdens {
struct snd_pcm_hw_constraint_list {
unsigned int count;
unsigned int *list;
const unsigned int *list;
unsigned int mask;
};
@ -454,6 +454,7 @@ struct snd_pcm {
void *private_data;
void (*private_free) (struct snd_pcm *pcm);
struct device *dev; /* actual hw device this belongs to */
bool internal; /* pcm is for internal use only */
#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
struct snd_pcm_oss oss;
#endif
@ -475,6 +476,9 @@ extern const struct file_operations snd_pcm_f_ops[2];
int snd_pcm_new(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
struct snd_pcm **rpcm);
int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
struct snd_pcm **rpcm);
int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count);
int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree);
@ -781,7 +785,8 @@ void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interva
unsigned int k, struct snd_interval *c);
void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
const struct snd_interval *b, struct snd_interval *c);
int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask);
int snd_interval_list(struct snd_interval *i, unsigned int count,
const unsigned int *list, unsigned int mask);
int snd_interval_ratnum(struct snd_interval *i,
unsigned int rats_count, struct snd_ratnum *rats,
unsigned int *nump, unsigned int *denp);

12
include/sound/sh_fsi.h
View File

@ -72,10 +72,16 @@
#define SH_FSI_BPFMD_32 (5 << 4)
#define SH_FSI_BPFMD_16 (6 << 4)
struct sh_fsi_port_info {
unsigned long flags;
int tx_id;
int rx_id;
int (*set_rate)(struct device *dev, int rate, int enable);
};
struct sh_fsi_platform_info {
unsigned long porta_flags;
unsigned long portb_flags;
int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
struct sh_fsi_port_info port_a;
struct sh_fsi_port_info port_b;
};
/*

11
include/sound/soc-dai.h
View File

@ -17,6 +17,7 @@
#include <linux/list.h>
struct snd_pcm_substream;
struct snd_soc_dapm_widget;
/*
* DAI hardware audio formats.
@ -238,6 +239,9 @@ struct snd_soc_dai {
unsigned char pop_wait:1;
unsigned char probed:1;
struct snd_soc_dapm_widget *playback_widget;
struct snd_soc_dapm_widget *capture_widget;
/* DAI DMA data */
void *playback_dma_data;
void *capture_dma_data;
@ -246,10 +250,9 @@ struct snd_soc_dai {
unsigned int rate;
/* parent platform/codec */
union {
struct snd_soc_platform *platform;
struct snd_soc_codec *codec;
};
struct snd_soc_platform *platform;
struct snd_soc_codec *codec;
struct snd_soc_card *card;
struct list_head list;

33
include/sound/soc-dapm.h
View File

@ -243,6 +243,10 @@
{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .event = wevent, \
.event_flags = wflags}
#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \
{ .id = snd_soc_dapm_regulator_supply, .name = wname, \
.reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
@ -322,6 +326,8 @@ struct snd_soc_dapm_context;
int dapm_reg_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
int dapm_regulator_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
/* dapm controls */
int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
@ -346,11 +352,12 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget);
int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget,
int num);
int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
struct snd_soc_dai *dai);
int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
/* dapm path setup */
int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
@ -361,10 +368,16 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
const char *stream, int event);
int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
struct snd_soc_dai *dai, int event);
void snd_soc_dapm_shutdown(struct snd_soc_card *card);
/* external DAPM widget events */
int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int connect);
int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
@ -411,9 +424,11 @@ enum snd_soc_dapm_type {
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
snd_soc_dapm_regulator_supply, /* external regulator */
snd_soc_dapm_aif_in, /* audio interface input */
snd_soc_dapm_aif_out, /* audio interface output */
snd_soc_dapm_siggen, /* signal generator */
snd_soc_dapm_dai, /* link to DAI structure */
};
/*
@ -434,8 +449,8 @@ struct snd_soc_dapm_route {
/* dapm audio path between two widgets */
struct snd_soc_dapm_path {
char *name;
char *long_name;
const char *name;
const char *long_name;
/* source (input) and sink (output) widgets */
struct snd_soc_dapm_widget *source;
@ -458,13 +473,15 @@ struct snd_soc_dapm_path {
/* dapm widget */
struct snd_soc_dapm_widget {
enum snd_soc_dapm_type id;
char *name; /* widget name */
char *sname; /* stream name */
const char *name; /* widget name */
const char *sname; /* stream name */
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct list_head list;
struct snd_soc_dapm_context *dapm;
void *priv; /* widget specific data */
/* dapm control */
short reg; /* negative reg = no direct dapm */
unsigned char shift; /* bits to shift */

45
include/sound/soc.h
View File

@ -185,6 +185,20 @@
.rreg = xreg_right, .shift = xshift, \
.min = xmin, .max = xmax} }
#define SND_SOC_BYTES(xname, xbase, xregs) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
.put = snd_soc_bytes_put, .private_value = \
((unsigned long)&(struct soc_bytes) \
{.base = xbase, .num_regs = xregs }) }
#define SND_SOC_BYTES_MASK(xname, xbase, xregs, xmask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
.put = snd_soc_bytes_put, .private_value = \
((unsigned long)&(struct soc_bytes) \
{.base = xbase, .num_regs = xregs, \
.mask = xmask }) }
/*
* Simplified versions of above macros, declaring a struct and calculating
@ -366,12 +380,16 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
*Controls
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
void *data, char *long_name,
void *data, const char *long_name,
const char *prefix);
int snd_soc_add_controls(struct snd_soc_codec *codec,
int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_card_controls(struct snd_soc_card *soc_card,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@ -409,6 +427,13 @@ int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
/**
* struct snd_soc_reg_access - Describes whether a given register is
@ -505,6 +530,7 @@ struct snd_soc_pcm_stream {
unsigned int rate_max; /* max rate */
unsigned int channels_min; /* min channels */
unsigned int channels_max; /* max channels */
unsigned int sig_bits; /* number of bits of content */
};
/* SoC audio ops */
@ -559,6 +585,7 @@ struct snd_soc_codec {
unsigned int ac97_created:1; /* Codec has been created by SoC */
unsigned int sysfs_registered:1; /* codec has been sysfs registered */
unsigned int cache_init:1; /* codec cache has been initialized */
unsigned int using_regmap:1; /* using regmap access */
u32 cache_only; /* Suppress writes to hardware */
u32 cache_sync; /* Cache needs to be synced to hardware */
@ -637,6 +664,8 @@ struct snd_soc_codec_driver {
/* codec stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@ -689,6 +718,7 @@ struct snd_soc_platform {
int id;
struct device *dev;
struct snd_soc_platform_driver *driver;
struct mutex mutex;
unsigned int suspended:1; /* platform is suspended */
unsigned int probed:1;
@ -698,6 +728,11 @@ struct snd_soc_platform {
struct list_head card_list;
struct snd_soc_dapm_context dapm;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_platform_root;
struct dentry *debugfs_dapm;
#endif
};
struct snd_soc_dai_link {
@ -875,6 +910,12 @@ struct soc_mixer_control {
unsigned int reg, rreg, shift, rshift, invert;
};
struct soc_bytes {
int base;
int num_regs;
u32 mask;
};
/* enumerated kcontrol */
struct soc_enum {
unsigned short reg;

2
include/sound/version.h
View File

@ -1,3 +1,3 @@
/* include/version.h */
#define CONFIG_SND_VERSION "1.0.24"
#define CONFIG_SND_VERSION "1.0.25"
#define CONFIG_SND_DATE ""

41
include/sound/wm2200.h Normal file
View File

@ -0,0 +1,41 @@
/*
* linux/sound/wm2200.h -- Platform data for WM2200
*
* Copyright 2012 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.
*/
#ifndef __LINUX_SND_WM2200_H
#define __LINUX_SND_WM2200_H
#define WM2200_GPIO_SET 0x10000
enum wm2200_in_mode {
WM2200_IN_SE = 0,
WM2200_IN_DIFF = 1,
WM2200_IN_DMIC = 2,
};
enum wm2200_dmic_sup {
WM2200_DMIC_SUP_MICVDD = 0,
WM2200_DMIC_SUP_MICBIAS1 = 1,
WM2200_DMIC_SUP_MICBIAS2 = 2,
};
struct wm2200_pdata {
int reset; /** GPIO controlling /RESET, if any */
int ldo_ena; /** GPIO controlling LODENA, if any */
int irq_flags;
int gpio_defaults[4];
enum wm2200_in_mode in_mode[3];
enum wm2200_dmic_sup dmic_sup[3];
int micbias_cfg[2]; /** Register value to configure MICBIAS */
};
#endif

6
include/sound/wm8962.h
View File

@ -49,6 +49,12 @@ struct wm8962_pdata {
bool irq_active_low;
bool spk_mono; /* Speaker outputs tied together as mono */
/**
* This flag should be set if one or both IN4 inputs is wired
* in a DC measurement configuration.
*/
bool in4_dc_measure;
};
#endif

2
include/sound/ymfpci.h
View File

@ -366,6 +366,8 @@ struct snd_ymfpci {
#ifdef CONFIG_PM
u32 *saved_regs;
u32 saved_ydsxgr_mode;
u16 saved_dsxg_legacy;
u16 saved_dsxg_elegacy;
#endif
};

13
sound/aoa/codecs/onyx.c
View File

@ -1132,15 +1132,4 @@ static struct i2c_driver onyx_driver = {
.id_table = onyx_i2c_id,
};
static int __init onyx_init(void)
{
return i2c_add_driver(&onyx_driver);
}
static void __exit onyx_exit(void)
{
i2c_del_driver(&onyx_driver);
}
module_init(onyx_init);
module_exit(onyx_exit);
module_i2c_driver(onyx_driver);

13
sound/aoa/codecs/tas.c
View File

@ -1026,15 +1026,4 @@ static struct i2c_driver tas_driver = {
.id_table = tas_i2c_id,
};
static int __init tas_init(void)
{
return i2c_add_driver(&tas_driver);
}
static void __exit tas_exit(void)
{
i2c_del_driver(&tas_driver);
}
module_init(tas_init);
module_exit(tas_exit);
module_i2c_driver(tas_driver);

2
sound/core/control.c
View File

@ -1313,7 +1313,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file,
err = -EPERM;
goto __kctl_end;
}
err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv);
err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv);
if (err > 0) {
up_read(&card->controls_rwsem);
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_TLV, &kctl->id);

169
sound/core/init.c
View File

@ -480,74 +480,104 @@ int snd_card_free(struct snd_card *card)
EXPORT_SYMBOL(snd_card_free);
static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid)
/* retrieve the last word of shortname or longname */
static const char *retrieve_id_from_card_name(const char *name)
{
int i, len, idx_flag = 0, loops = SNDRV_CARDS;
const char *spos, *src;
const char *spos = name;
while (*name) {
if (isspace(*name) && isalnum(name[1]))
spos = name + 1;
name++;
}
return spos;
}
/* return true if the given id string doesn't conflict any other card ids */
static bool card_id_ok(struct snd_card *card, const char *id)
{
int i;
if (!snd_info_check_reserved_words(id))
return false;
for (i = 0; i < snd_ecards_limit; i++) {
if (snd_cards[i] && snd_cards[i] != card &&
!strcmp(snd_cards[i]->id, id))
return false;
}
return true;
}
/* copy to card->id only with valid letters from nid */
static void copy_valid_id_string(struct snd_card *card, const char *src,
const char *nid)
{
char *id = card->id;
while (*nid && !isalnum(*nid))
nid++;
if (isdigit(*nid))
*id++ = isalpha(*src) ? *src : 'D';
while (*nid && (size_t)(id - card->id) < sizeof(card->id) - 1) {
if (isalnum(*nid))
*id++ = *nid;
nid++;
}
*id = 0;
}
/* Set card->id from the given string
* If the string conflicts with other ids, add a suffix to make it unique.
*/
static void snd_card_set_id_no_lock(struct snd_card *card, const char *src,
const char *nid)
{
int len, loops;
bool with_suffix;
bool is_default = false;
char *id;
if (nid == NULL) {
id = card->shortname;
spos = src = id;
while (*id != '\0') {
if (*id == ' ')
spos = id + 1;
id++;
}
} else {
spos = src = nid;
}
copy_valid_id_string(card, src, nid);
id = card->id;
while (*spos != '\0' && !isalnum(*spos))
spos++;
if (isdigit(*spos))
*id++ = isalpha(src[0]) ? src[0] : 'D';
while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) {
if (isalnum(*spos))
*id++ = *spos;
spos++;
}
*id = '\0';
id = card->id;
if (*id == '\0')
again:
/* use "Default" for obviously invalid strings
* ("card" conflicts with proc directories)
*/
if (!*id || !strncmp(id, "card", 4)) {
strcpy(id, "Default");
is_default = true;
}
while (1) {
if (loops-- == 0) {
snd_printk(KERN_ERR "unable to set card id (%s)\n", id);
strcpy(card->id, card->proc_root->name);
return;
}
if (!snd_info_check_reserved_words(id))
goto __change;
for (i = 0; i < snd_ecards_limit; i++) {
if (snd_cards[i] && !strcmp(snd_cards[i]->id, id))
goto __change;
}
break;
with_suffix = false;
for (loops = 0; loops < SNDRV_CARDS; loops++) {
if (card_id_ok(card, id))
return; /* OK */
__change:
len = strlen(id);
if (idx_flag) {
if (id[len-1] != '9')
id[len-1]++;
else
id[len-1] = 'A';
} else if ((size_t)len <= sizeof(card->id) - 3) {
strcat(id, "_1");
idx_flag++;
if (!with_suffix) {
/* add the "_X" suffix */
char *spos = id + len;
if (len > sizeof(card->id) - 3)
spos = id + sizeof(card->id) - 3;
strcpy(spos, "_1");
with_suffix = true;
} else {
spos = id + len - 2;
if ((size_t)len <= sizeof(card->id) - 2)
spos++;
*(char *)spos++ = '_';
*(char *)spos++ = '1';
*(char *)spos++ = '\0';
idx_flag++;
/* modify the existing suffix */
if (id[len - 1] != '9')
id[len - 1]++;
else
id[len - 1] = 'A';
}
}
/* fallback to the default id */
if (!is_default) {
*id = 0;
goto again;
}
/* last resort... */
snd_printk(KERN_ERR "unable to set card id (%s)\n", id);
if (card->proc_root->name)
strcpy(card->id, card->proc_root->name);
}
/**
@ -564,7 +594,7 @@ void snd_card_set_id(struct snd_card *card, const char *nid)
if (card->id[0] != '\0')
return;
mutex_lock(&snd_card_mutex);
snd_card_set_id_no_lock(card, nid);
snd_card_set_id_no_lock(card, nid, nid);
mutex_unlock(&snd_card_mutex);
}
EXPORT_SYMBOL(snd_card_set_id);
@ -596,22 +626,12 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr,
memcpy(buf1, buf, copy);
buf1[copy] = '\0';
mutex_lock(&snd_card_mutex);
if (!snd_info_check_reserved_words(buf1)) {
__exist:
if (!card_id_ok(NULL, buf1)) {
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)) {
if (card == snd_cards[idx])
goto __ok;
else
goto __exist;
}
}
strcpy(card->id, buf1);
snd_info_card_id_change(card);
__ok:
mutex_unlock(&snd_card_mutex);
return count;
@ -665,7 +685,18 @@ int snd_card_register(struct snd_card *card)
mutex_unlock(&snd_card_mutex);
return 0;
}
snd_card_set_id_no_lock(card, card->id[0] == '\0' ? NULL : card->id);
if (*card->id) {
/* make a unique id name from the given string */
char tmpid[sizeof(card->id)];
memcpy(tmpid, card->id, sizeof(card->id));
snd_card_set_id_no_lock(card, tmpid, tmpid);
} else {
/* create an id from either shortname or longname */
const char *src;
src = *card->shortname ? card->shortname : card->longname;
snd_card_set_id_no_lock(card, src,
retrieve_id_from_card_name(src));
}
snd_cards[card->number] = card;
mutex_unlock(&snd_card_mutex);
init_info_for_card(card);

4
sound/core/jack.c
View File

@ -25,7 +25,7 @@
#include <sound/jack.h>
#include <sound/core.h>
static int jack_switch_types[] = {
static int jack_switch_types[SND_JACK_SWITCH_TYPES] = {
SW_HEADPHONE_INSERT,
SW_MICROPHONE_INSERT,
SW_LINEOUT_INSERT,
@ -128,7 +128,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
jack->type = type;
for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++)
for (i = 0; i < SND_JACK_SWITCH_TYPES; i++)
if (type & (1 << i))
input_set_capability(jack->input_dev, EV_SW,
jack_switch_types[i]);

2
sound/core/misc.c
View File

@ -72,7 +72,7 @@ void __snd_printk(unsigned int level, const char *path, int line,
char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV";
#endif
#ifdef CONFIG_SND_DEBUG
#ifdef CONFIG_SND_DEBUG
if (debug < level)
return;
#endif

99
sound/core/pcm.c
View File

@ -650,7 +650,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
pstr->stream = stream;
pstr->pcm = pcm;
pstr->substream_count = substream_count;
if (substream_count > 0) {
if (substream_count > 0 && !pcm->internal) {
err = snd_pcm_stream_proc_init(pstr);
if (err < 0) {
snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
@ -674,15 +674,18 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
pstr->substream = substream;
else
prev->next = substream;
err = snd_pcm_substream_proc_init(substream);
if (err < 0) {
snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
if (prev == NULL)
pstr->substream = NULL;
else
prev->next = NULL;
kfree(substream);
return err;
if (!pcm->internal) {
err = snd_pcm_substream_proc_init(substream);
if (err < 0) {
snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
if (prev == NULL)
pstr->substream = NULL;
else
prev->next = NULL;
kfree(substream);
return err;
}
}
substream->group = &substream->self_group;
spin_lock_init(&substream->self_group.lock);
@ -696,25 +699,9 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
EXPORT_SYMBOL(snd_pcm_new_stream);
/**
* snd_pcm_new - create a new PCM instance
* @card: the card instance
* @id: the id string
* @device: the device index (zero based)
* @playback_count: the number of substreams for playback
* @capture_count: the number of substreams for capture
* @rpcm: the pointer to store the new pcm instance
*
* Creates a new PCM instance.
*
* The pcm operators have to be set afterwards to the new instance
* via snd_pcm_set_ops().
*
* Returns zero if successful, or a negative error code on failure.
*/
int snd_pcm_new(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
struct snd_pcm ** rpcm)
static int _snd_pcm_new(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count, bool internal,
struct snd_pcm **rpcm)
{
struct snd_pcm *pcm;
int err;
@ -735,6 +722,7 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device,
}
pcm->card = card;
pcm->device = device;
pcm->internal = internal;
if (id)
strlcpy(pcm->id, id, sizeof(pcm->id));
if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) {
@ -756,8 +744,59 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device,
return 0;
}
/**
* snd_pcm_new - create a new PCM instance
* @card: the card instance
* @id: the id string
* @device: the device index (zero based)
* @playback_count: the number of substreams for playback
* @capture_count: the number of substreams for capture
* @rpcm: the pointer to store the new pcm instance
*
* Creates a new PCM instance.
*
* The pcm operators have to be set afterwards to the new instance
* via snd_pcm_set_ops().
*
* Returns zero if successful, or a negative error code on failure.
*/
int snd_pcm_new(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count, struct snd_pcm **rpcm)
{
return _snd_pcm_new(card, id, device, playback_count, capture_count,
false, rpcm);
}
EXPORT_SYMBOL(snd_pcm_new);
/**
* snd_pcm_new_internal - create a new internal PCM instance
* @card: the card instance
* @id: the id string
* @device: the device index (zero based - shared with normal PCMs)
* @playback_count: the number of substreams for playback
* @capture_count: the number of substreams for capture
* @rpcm: the pointer to store the new pcm instance
*
* Creates a new internal PCM instance with no userspace device or procfs
* entries. This is used by ASoC Back End PCMs in order to create a PCM that
* will only be used internally by kernel drivers. i.e. it cannot be opened
* by userspace. It provides existing ASoC components drivers with a substream
* and access to any private data.
*
* The pcm operators have to be set afterwards to the new instance
* via snd_pcm_set_ops().
*
* Returns zero if successful, or a negative error code on failure.
*/
int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
struct snd_pcm **rpcm)
{
return _snd_pcm_new(card, id, device, playback_count, capture_count,
true, rpcm);
}
EXPORT_SYMBOL(snd_pcm_new_internal);
static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
{
struct snd_pcm_substream *substream, *substream_next;
@ -994,7 +1033,7 @@ static int snd_pcm_dev_register(struct snd_device *device)
}
for (cidx = 0; cidx < 2; cidx++) {
int devtype = -1;
if (pcm->streams[cidx].substream == NULL)
if (pcm->streams[cidx].substream == NULL || pcm->internal)
continue;
switch (cidx) {
case SNDRV_PCM_STREAM_PLAYBACK:

3
sound/core/pcm_lib.c
View File

@ -1029,7 +1029,8 @@ static int snd_interval_ratden(struct snd_interval *i,
*
* Returns non-zero if the value is changed, zero if not changed.
*/
int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask)
int snd_interval_list(struct snd_interval *i, unsigned int count,
const unsigned int *list, unsigned int mask)
{
unsigned int k;
struct snd_interval list_range;

15
sound/core/pcm_native.c
View File

@ -1586,12 +1586,18 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
struct file *file;
struct snd_pcm_file *pcm_file;
struct snd_pcm_substream *substream1;
struct snd_pcm_group *group;
file = snd_pcm_file_fd(fd);
if (!file)
return -EBADFD;
pcm_file = file->private_data;
substream1 = pcm_file->substream;
group = kmalloc(sizeof(*group), GFP_KERNEL);
if (!group) {
res = -ENOMEM;
goto _nolock;
}
down_write(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
@ -1604,11 +1610,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
goto _end;
}
if (!snd_pcm_stream_linked(substream)) {
substream->group = kmalloc(sizeof(struct snd_pcm_group), GFP_ATOMIC);
if (substream->group == NULL) {
res = -ENOMEM;
goto _end;
}
substream->group = group;
spin_lock_init(&substream->group->lock);
INIT_LIST_HEAD(&substream->group->substreams);
list_add_tail(&substream->link_list, &substream->group->substreams);
@ -1620,7 +1622,10 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
_end:
write_unlock_irq(&snd_pcm_link_rwlock);
up_write(&snd_pcm_link_rwsem);
_nolock:
fput(file);
if (res < 0)
kfree(group);
return res;
}

46
sound/core/vmaster.c
View File

@ -37,6 +37,8 @@ struct link_master {
struct link_ctl_info info;
int val; /* the master value */
unsigned int tlv[4];
void (*hook)(void *private_data, int);
void *hook_private_data;
};
/*
@ -126,7 +128,9 @@ static int master_init(struct link_master *master)
master->info.count = 1; /* always mono */
/* set full volume as default (= no attenuation) */
master->val = master->info.max_val;
return 0;
if (master->hook)
master->hook(master->hook_private_data, master->val);
return 1;
}
return -ENOENT;
}
@ -329,6 +333,8 @@ static int master_put(struct snd_kcontrol *kcontrol,
slave_put_val(slave, uval);
}
kfree(uval);
if (master->hook && !err)
master->hook(master->hook_private_data, master->val);
return 1;
}
@ -408,3 +414,41 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
return kctl;
}
EXPORT_SYMBOL(snd_ctl_make_virtual_master);
/**
* snd_ctl_add_vmaster_hook - Add a hook to a vmaster control
* @kcontrol: vmaster kctl element
* @hook: the hook function
*
* Adds the given hook to the vmaster control element so that it's called
* at each time when the value is changed.
*/
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol,
void (*hook)(void *private_data, int),
void *private_data)
{
struct link_master *master = snd_kcontrol_chip(kcontrol);
master->hook = hook;
master->hook_private_data = private_data;
return 0;
}
EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook);
/**
* snd_ctl_sync_vmaster_hook - Sync the vmaster hook
* @kcontrol: vmaster kctl element
*
* Call the hook function to synchronize with the current value of the given
* vmaster element. NOP when NULL is passed to @kcontrol or the hook doesn't
* exist.
*/
void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kcontrol)
{
struct link_master *master;
if (!kcontrol)
return;
master = snd_kcontrol_chip(kcontrol);
if (master->hook)
master->hook(master->hook_private_data, master->val);
}
EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster_hook);

13
sound/pci/au88x0/au88x0.h
View File

@ -26,7 +26,7 @@
#include <sound/mpu401.h>
#include <sound/hwdep.h>
#include <sound/ac97_codec.h>
#include <sound/tlv.h>
#endif
#ifndef CHIP_AU8820
@ -107,6 +107,14 @@
#define NR_WTPB 0x20 /* WT channels per each bank. */
#define NR_PCM 0x10
struct pcm_vol {
struct snd_kcontrol *kctl;
int active;
int dma;
int mixin[4];
int vol[4];
};
/* Structs */
typedef struct {
//int this_08; /* Still unknown */
@ -168,6 +176,7 @@ struct snd_vortex {
/* Xtalk canceler */
int xt_mode; /* 1: speakers, 0:headphones. */
#endif
struct pcm_vol pcm_vol[NR_PCM];
int isquad; /* cache of extended ID codec flag. */
@ -239,7 +248,7 @@ static int vortex_alsafmt_aspfmt(int alsafmt);
/* Connection stuff. */
static void vortex_connect_default(vortex_t * vortex, int en);
static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch,
int dir, int type);
int dir, int type, int subdev);
static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out,
int restype);
#ifndef CHIP_AU8810

20
sound/pci/au88x0/au88x0_core.c
View File

@ -2050,8 +2050,6 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
}
/* Default Connections */
static int
vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type);
static void vortex_connect_default(vortex_t * vortex, int en)
{
@ -2111,15 +2109,13 @@ static void vortex_connect_default(vortex_t * vortex, int en)
Return: Return allocated DMA or same DMA passed as "dma" when dma >= 0.
*/
static int
vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type)
vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
int type, int subdev)
{
stream_t *stream;
int i, en;
struct pcm_vol *p;
if ((nr_ch == 3)
|| ((dir == SNDRV_PCM_STREAM_CAPTURE) && (nr_ch > 2)))
return -EBUSY;
if (dma >= 0) {
en = 0;
vortex_adb_checkinout(vortex,
@ -2250,6 +2246,14 @@ vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type)
MIX_DEFIGAIN);
#endif
}
if (stream->type == VORTEX_PCM_ADB && en) {
p = &vortex->pcm_vol[subdev];
p->dma = dma;
for (i = 0; i < nr_ch; i++)
p->mixin[i] = mix[i];
for (i = 0; i < ch_top; i++)
p->vol[i] = 0;
}
}
#ifndef CHIP_AU8820
else {
@ -2473,7 +2477,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id)
hwread(vortex->mmio, VORTEX_IRQ_STAT);
handled = 1;
}
if (source & IRQ_MIDI) {
if ((source & IRQ_MIDI) && vortex->rmidi) {
snd_mpu401_uart_interrupt(vortex->irq,
vortex->rmidi->private_data);
handled = 1;

127
sound/pci/au88x0/au88x0_pcm.c
View File

@ -122,6 +122,18 @@ static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = {
.mask = 0,
};
#endif
static void vortex_notify_pcm_vol_change(struct snd_card *card,
struct snd_kcontrol *kctl, int activate)
{
if (activate)
kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
else
kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE |
SNDRV_CTL_EVENT_MASK_INFO, &(kctl->id));
}
/* open callback */
static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
{
@ -230,12 +242,14 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream,
if (stream != NULL)
vortex_adb_allocroute(chip, stream->dma,
stream->nr_ch, stream->dir,
stream->type);
stream->type,
substream->number);
/* Alloc routes. */
dma =
vortex_adb_allocroute(chip, -1,
params_channels(hw_params),
substream->stream, type);
substream->stream, type,
substream->number);
if (dma < 0) {
spin_unlock_irq(&chip->lock);
return dma;
@ -246,6 +260,11 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream,
vortex_adbdma_setbuffers(chip, dma,
params_period_bytes(hw_params),
params_periods(hw_params));
if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) {
chip->pcm_vol[substream->number].active = 1;
vortex_notify_pcm_vol_change(chip->card,
chip->pcm_vol[substream->number].kctl, 1);
}
}
#ifndef CHIP_AU8810
else {
@ -275,10 +294,18 @@ static int snd_vortex_pcm_hw_free(struct snd_pcm_substream *substream)
spin_lock_irq(&chip->lock);
// Delete audio routes.
if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
if (stream != NULL)
if (stream != NULL) {
if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) {
chip->pcm_vol[substream->number].active = 0;
vortex_notify_pcm_vol_change(chip->card,
chip->pcm_vol[substream->number].kctl,
0);
}
vortex_adb_allocroute(chip, stream->dma,
stream->nr_ch, stream->dir,
stream->type);
stream->type,
substream->number);
}
}
#ifndef CHIP_AU8810
else {
@ -506,6 +533,83 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = {
},
};
/* subdevice PCM Volume control */
static int snd_vortex_pcm_vol_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
vortex_t *vortex = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = (VORTEX_IS_QUAD(vortex) ? 4 : 2);
uinfo->value.integer.min = -128;
uinfo->value.integer.max = 32;
return 0;
}
static int snd_vortex_pcm_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int i;
vortex_t *vortex = snd_kcontrol_chip(kcontrol);
int subdev = kcontrol->id.subdevice;
struct pcm_vol *p = &vortex->pcm_vol[subdev];
int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2);
for (i = 0; i < max_chn; i++)
ucontrol->value.integer.value[i] = p->vol[i];
return 0;
}
static int snd_vortex_pcm_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int i;
int changed = 0;
int mixin;
unsigned char vol;
vortex_t *vortex = snd_kcontrol_chip(kcontrol);
int subdev = kcontrol->id.subdevice;
struct pcm_vol *p = &vortex->pcm_vol[subdev];
int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2);
for (i = 0; i < max_chn; i++) {
if (p->vol[i] != ucontrol->value.integer.value[i]) {
p->vol[i] = ucontrol->value.integer.value[i];
if (p->active) {
switch (vortex->dma_adb[p->dma].nr_ch) {
case 1:
mixin = p->mixin[0];
break;
case 2:
default:
mixin = p->mixin[(i < 2) ? i : (i - 2)];
break;
case 4:
mixin = p->mixin[i];
break;
};
vol = p->vol[i];
vortex_mix_setinputvolumebyte(vortex,
vortex->mixplayb[i], mixin, vol);
}
changed = 1;
}
}
return changed;
}
static const DECLARE_TLV_DB_MINMAX(vortex_pcm_vol_db_scale, -9600, 2400);
static struct snd_kcontrol_new snd_vortex_pcm_vol __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = "PCM Playback Volume",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
SNDRV_CTL_ELEM_ACCESS_TLV_READ |
SNDRV_CTL_ELEM_ACCESS_INACTIVE,
.info = snd_vortex_pcm_vol_info,
.get = snd_vortex_pcm_vol_get,
.put = snd_vortex_pcm_vol_put,
.tlv = { .p = vortex_pcm_vol_db_scale },
};
/* create a pcm device */
static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
{
@ -555,5 +659,20 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
return err;
}
}
if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_ADB) {
for (i = 0; i < NR_PCM; i++) {
chip->pcm_vol[i].active = 0;
chip->pcm_vol[i].dma = -1;
kctl = snd_ctl_new1(&snd_vortex_pcm_vol, chip);
if (!kctl)
return -ENOMEM;
chip->pcm_vol[i].kctl = kctl;
kctl->id.device = 0;
kctl->id.subdevice = i;
err = snd_ctl_add(chip->card, kctl);
if (err < 0)
return err;
}
}
return 0;
}

2
sound/pci/ctxfi/ctvmem.c
View File

@ -36,7 +36,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size)
size = CT_PAGE_ALIGN(size);
if (size > vm->size) {
printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural "
printk(KERN_ERR "ctxfi: Fail! No sufficient device virtual "
"memory space available!\n");
return NULL;
}

968
sound/pci/hda/alc260_quirks.c
View File

@ -1,968 +0,0 @@
/*
* ALC260 quirk models
* included by patch_realtek.c
*/
/* ALC260 models */
enum {
ALC260_AUTO,
ALC260_BASIC,
ALC260_FUJITSU_S702X,
ALC260_ACER,
ALC260_WILL,
ALC260_REPLACER_672V,
ALC260_FAVORIT100,
#ifdef CONFIG_SND_DEBUG
ALC260_TEST,
#endif
ALC260_MODEL_LAST /* last tag */
};
static const hda_nid_t alc260_dac_nids[1] = {
/* front */
0x02,
};
static const hda_nid_t alc260_adc_nids[1] = {
/* ADC0 */
0x04,
};
static const hda_nid_t alc260_adc_nids_alt[1] = {
/* ADC1 */
0x05,
};
/* NIDs used when simultaneous access to both ADCs makes sense. Note that
* alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
*/
static const hda_nid_t alc260_dual_adc_nids[2] = {
/* ADC0, ADC1 */
0x04, 0x05
};
#define ALC260_DIGOUT_NID 0x03
#define ALC260_DIGIN_NID 0x06
static const struct hda_input_mux alc260_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
* headphone jack and the internal CD lines since these are the only pins at
* which audio can appear. For flexibility, also allow the option of
* recording the mixer output on the second ADC (ADC0 doesn't have a
* connection to the mixer output).
*/
static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
{
.num_items = 3,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
},
},
{
.num_items = 4,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
{ "Mixer", 0x5 },
},
},
};
/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
* the Fujitsu S702x, but jacks are marked differently.
*/
static const struct hda_input_mux alc260_acer_capture_sources[2] = {
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x5 },
},
},
{
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x6 },
{ "Mixer", 0x5 },
},
},
};
/* Maxdata Favorit 100XS */
static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
{
.num_items = 2,
.items = {
{ "Line/Mic", 0x0 },
{ "CD", 0x4 },
},
},
{
.num_items = 3,
.items = {
{ "Line/Mic", 0x0 },
{ "CD", 0x4 },
{ "Mixer", 0x5 },
},
},
};
/*
* This is just place-holder, so there's something for alc_build_pcms to look
* at when it calculates the maximum number of channels. ALC260 has no mixer
* element which allows changing the channel mode, so the verb list is
* never used.
*/
static const struct hda_channel_mode alc260_modes[1] = {
{ 2, NULL },
};
/* Mixer combinations
*
* basic: base_output + input + pc_beep + capture
* fujitsu: fujitsu + capture
* acer: acer + capture
*/
static const struct snd_kcontrol_new alc260_base_output_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc260_input_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
{ } /* end */
};
/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
* HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
*/
static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
{ } /* end */
};
/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
* versions of the ALC260 don't act on requests to enable mic bias from NID
* 0x0f (used to drive the headphone jack in these laptops). The ALC260
* datasheet doesn't mention this restriction. At this stage it's not clear
* whether this behaviour is intentional or is a hardware bug in chip
* revisions available in early 2006. Therefore for now allow the
* "Headphone Jack Mode" control to span all choices, but if it turns out
* that the lack of mic bias for this NID is intentional we could change the
* mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*
* In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
* don't appear to make the mic bias available from the "line" jack, even
* though the NID used for this jack (0x14) can supply it. The theory is
* that perhaps Acer have included blocking capacitors between the ALC260
* and the output jack. If this turns out to be the case for all such
* models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
* to ALC_PIN_DIR_INOUT_NOMICBIAS.
*
* The C20x Tablet series have a mono internal speaker which is controlled
* via the chip's Mono sum widget and pin complex, so include the necessary
* controls for such models. On models without a "mono speaker" the control
* won't do anything.
*/
static const struct snd_kcontrol_new alc260_acer_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
{ } /* end */
};
/* Maxdata Favorit 100XS: one output and one input (0x12) jack
*/
static const struct snd_kcontrol_new alc260_favorit100_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
{ } /* end */
};
/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
* Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
*/
static const struct snd_kcontrol_new alc260_will_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
{ } /* end */
};
/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
* Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
*/
static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
{ } /* end */
};
/*
* initialization verbs
*/
static const struct hda_verb alc260_init_verbs[] = {
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* LINE-2 is used for line-out in rear */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* select line-out */
{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LINE-OUT pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable HP */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* enable Mono */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* mute capture amp left and right */
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
/* set vol=0 Line-Out mixer amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 HP mixer amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 Mono mixer amp left and right */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* unmute LINE-2 out pin */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
/* mute analog inputs */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* mute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
* laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
* audio = 0x16, internal speaker = 0x10.
*/
static const struct hda_verb alc260_fujitsu_init_verbs[] = {
/* Disable all GPIOs */
{0x01, AC_VERB_SET_GPIO_MASK, 0},
/* Internal speaker is connected to headphone pin */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Headphone/Line-out jack connects to Line1 pin; make it an output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Mic/Line-in jack is connected to mic1 pin, so make it an input */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Ensure all other unused pins are disabled and muted. */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Line1 pin widget takes its input from the OUT1 sum bus
* when acting as an output.
*/
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Line1 pin widget output buffer since it starts as an output.
* If the pin mode is changed by the user the pin mode control will
* take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute input buffer of pin widget used for Line-in (no equiv
* mixer ctrl)
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - line
* in (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to line in (on mic1 pin)
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
* similar laptops (adapted from Fujitsu init verbs).
*/
static const struct hda_verb alc260_acer_init_verbs[] = {
/* On TravelMate laptops, GPIO 0 enables the internal speaker and
* the headphone jack. Turn this on and rely on the standard mute
* methods whenever the user wants to turn these outputs off.
*/
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
/* Internal speaker/Headphone jack is connected to Line-out pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Internal microphone/Mic jack is connected to Mic1 pin */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
/* Line In jack is connected to Line1 pin */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Ensure all other unused pins are disabled and muted. */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
* bus when acting as outputs.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute Line-out pin widget amp left and right
* (no equiv mixer ctrl)
*/
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute mono pin widget amp output (no equiv mixer ctrl) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mic1 and Line1 pin widget input buffers since they start as
* inputs. If the pin mode is changed by the user the pin mode control
* will take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - mic
* (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do similar with the second ADC: mute capture input amp and
* set ADC connection to mic to match ALSA's default state.
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Initialisation sequence for Maxdata Favorit 100XS
* (adapted from Acer init verbs).
*/
static const struct hda_verb alc260_favorit100_init_verbs[] = {
/* GPIO 0 enables the output jack.
* Turn this on and rely on the standard mute
* methods whenever the user wants to turn these outputs off.
*/
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
/* Line/Mic input jack is connected to Mic1 pin */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
/* Ensure all other unused pins are disabled and muted. */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
* bus when acting as outputs.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute Line-out pin widget amp left and right
* (no equiv mixer ctrl)
*/
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mic1 and Line1 pin widget input buffers since they start as
* inputs. If the pin mode is changed by the user the pin mode control
* will take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - mic
* (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do similar with the second ADC: mute capture input amp and
* set ADC connection to mic to match ALSA's default state.
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
static const struct hda_verb alc260_will_verbs[] = {
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
{0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
{}
};
static const struct hda_verb alc260_replacer_672v_verbs[] = {
{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
{0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
/* toggle speaker-output according to the hp-jack state */
static void alc260_replacer_672v_automute(struct hda_codec *codec)
{
unsigned int present;
/* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
present = snd_hda_jack_detect(codec, 0x0f);
if (present) {
snd_hda_codec_write_cache(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 1);
snd_hda_codec_write_cache(codec, 0x0f, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
PIN_HP);
} else {
snd_hda_codec_write_cache(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 0);
snd_hda_codec_write_cache(codec, 0x0f, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
PIN_OUT);
}
}
static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc260_replacer_672v_automute(codec);
}
static const struct hda_verb alc260_hp_dc7600_verbs[] = {
{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
/* Test configuration for debugging, modelled after the ALC880 test
* configuration.
*/
#ifdef CONFIG_SND_DEBUG
static const hda_nid_t alc260_test_dac_nids[1] = {
0x02,
};
static const hda_nid_t alc260_test_adc_nids[2] = {
0x04, 0x05,
};
/* For testing the ALC260, each input MUX needs its own definition since
* the signal assignments are different. This assumes that the first ADC
* is NID 0x04.
*/
static const struct hda_input_mux alc260_test_capture_sources[2] = {
{
.num_items = 7,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "LINE-OUT pin", 0x5 },
{ "HP-OUT pin", 0x6 },
},
},
{
.num_items = 8,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "Mixer", 0x5 },
{ "LINE-OUT pin", 0x6 },
{ "HP-OUT pin", 0x7 },
},
},
};
static const struct snd_kcontrol_new alc260_test_mixer[] = {
/* Output driver widgets */
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
/* Modes for retasking pin widgets
* Note: the ALC260 doesn't seem to act on requests to enable mic
* bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
* mention this restriction. At this stage it's not clear whether
* this behaviour is intentional or is a hardware bug in chip
* revisions available at least up until early 2006. Therefore for
* now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
* choices, but if it turns out that the lack of mic bias for these
* NIDs is intentional we could change their modes from
* ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*/
ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
/* Loopback mixer controls */
HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
/* Controls for GPIO pins, assuming they are configured as outputs */
ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
/* Switches to allow the digital IO pins to be enabled. The datasheet
* is ambigious as to which NID is which; testing on laptops which
* make this output available should provide clarification.
*/
ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
/* A switch allowing EAPD to be enabled. Some laptops seem to use
* this output to turn on an external amplifier.
*/
ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
{ } /* end */
};
static const struct hda_verb alc260_test_init_verbs[] = {
/* Enable all GPIOs as outputs with an initial value of 0 */
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
{0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
/* Enable retasking pins as output, initially without power amp */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Disable digital (SPDIF) pins initially, but users can enable
* them via a mixer switch. In the case of SPDIF-out, this initverb
* payload also sets the generation to 0, output to be in "consumer"
* PCM format, copyright asserted, no pre-emphasis and no validity
* control.
*/
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
* OUT1 sum bus when acting as an output.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0c, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
{0x0e, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute retasking pin widget output buffers since the default
* state appears to be output. As the pin mode is changed by the
* user the pin mode control will take care of enabling the pin's
* input/output buffers as needed.
*/
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Also unmute the mono-out pin widget */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting (mic1
* pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to mic1 pin
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
#endif
/*
* ALC260 configurations
*/
static const char * const alc260_models[ALC260_MODEL_LAST] = {
[ALC260_BASIC] = "basic",
[ALC260_FUJITSU_S702X] = "fujitsu",
[ALC260_ACER] = "acer",
[ALC260_WILL] = "will",
[ALC260_REPLACER_672V] = "replacer",
[ALC260_FAVORIT100] = "favorit100",
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = "test",
#endif
[ALC260_AUTO] = "auto",
};
static const struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
{}
};
static const struct alc_config_preset alc260_presets[] = {
[ALC260_BASIC] = {
.mixers = { alc260_base_output_mixer,
alc260_input_mixer },
.init_verbs = { alc260_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_FUJITSU_S702X] = {
.mixers = { alc260_fujitsu_mixer },
.init_verbs = { alc260_fujitsu_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
.input_mux = alc260_fujitsu_capture_sources,
},
[ALC260_ACER] = {
.mixers = { alc260_acer_mixer },
.init_verbs = { alc260_acer_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
.input_mux = alc260_acer_capture_sources,
},
[ALC260_FAVORIT100] = {
.mixers = { alc260_favorit100_mixer },
.init_verbs = { alc260_favorit100_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
.input_mux = alc260_favorit100_capture_sources,
},
[ALC260_WILL] = {
.mixers = { alc260_will_mixer },
.init_verbs = { alc260_init_verbs, alc260_will_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
.adc_nids = alc260_adc_nids,
.dig_out_nid = ALC260_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_REPLACER_672V] = {
.mixers = { alc260_replacer_672v_mixer },
.init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
.adc_nids = alc260_adc_nids,
.dig_out_nid = ALC260_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
.unsol_event = alc260_replacer_672v_unsol_event,
.init_hook = alc260_replacer_672v_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = {
.mixers = { alc260_test_mixer },
.init_verbs = { alc260_test_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
.dac_nids = alc260_test_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
.adc_nids = alc260_test_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
.input_mux = alc260_test_capture_sources,
},
#endif
};

1707
sound/pci/hda/alc880_quirks.c
View File

File diff suppressed because it is too large Load Diff

866
sound/pci/hda/alc882_quirks.c
View File

@ -1,866 +0,0 @@
/*
* ALC882/ALC883/ALC888/ALC889 quirk models
* included by patch_realtek.c
*/
/* ALC882 models */
enum {
ALC882_AUTO,
ALC885_MBA21,
ALC885_MBP3,
ALC885_MB5,
ALC885_MACMINI3,
ALC885_IMAC91,
ALC889A_MB31,
ALC882_MODEL_LAST,
};
#define ALC882_DIGOUT_NID 0x06
#define ALC882_DIGIN_NID 0x0a
#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID
#define ALC883_DIGIN_NID ALC882_DIGIN_NID
#define ALC1200_DIGOUT_NID 0x10
static const struct hda_channel_mode alc882_ch_modes[1] = {
{ 8, NULL }
};
/* DACs */
static const hda_nid_t alc882_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
#define alc883_dac_nids alc882_dac_nids
/* ADCs */
#define alc882_adc_nids alc880_adc_nids
#define alc882_adc_nids_alt alc880_adc_nids_alt
#define alc883_adc_nids alc882_adc_nids_alt
static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
#define alc883_capsrc_nids alc882_capsrc_nids_alt
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static const struct hda_input_mux alc882_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
#define alc883_capture_source alc882_capture_source
static const struct hda_input_mux mb5_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x7 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux macmini3_capture_source = {
.num_items = 2,
.items = {
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux alc883_3stack_6ch_intel = {
.num_items = 4,
.items = {
{ "Mic", 0x1 },
{ "Front Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux alc889A_mb31_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
/* Front Mic (0x01) unused */
{ "Line", 0x2 },
/* Line 2 (0x03) unused */
/* CD (0x04) unused? */
},
};
static const struct hda_input_mux alc889A_imac91_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x01 },
{ "Line", 0x2 }, /* Not sure! */
},
};
/* Macbook Air 2,1 */
static const struct hda_channel_mode alc885_mba21_ch_modes[1] = {
{ 2, NULL },
};
/*
* macbook pro ALC885 can switch LineIn to LineOut without losing Mic
*/
/*
* 2ch mode
*/
static const struct hda_verb alc885_mbp_ch2_init[] = {
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ } /* end */
};
/*
* 4ch mode
*/
static const struct hda_verb alc885_mbp_ch4_init[] = {
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{ } /* end */
};
static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
{ 2, alc885_mbp_ch2_init },
{ 4, alc885_mbp_ch4_init },
};
/*
* 2ch
* Speakers/Woofer/HP = Front
* LineIn = Input
*/
static const struct hda_verb alc885_mb5_ch2_init[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ } /* end */
};
/*
* 6ch mode
* Speakers/HP = Front
* Woofer = LFE
* LineIn = Surround
*/
static const struct hda_verb alc885_mb5_ch6_init[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{ } /* end */
};
static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
{ 2, alc885_mb5_ch2_init },
{ 6, alc885_mb5_ch6_init },
};
#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes
/* Macbook Air 2,1 same control for HP and internal Speaker */
static const struct snd_kcontrol_new alc885_mba21_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT),
{ }
};
static const struct snd_kcontrol_new alc885_mbp3_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc885_mb5_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc885_macmini3_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc885_imac91_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc882_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static const struct hda_verb alc882_base_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* CLFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Side mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Rear Pin: output 1 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* CLFE Pin: output 2 (0x0e) */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Side Pin: output 3 (0x0f) */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
#define alc883_init_verbs alc882_base_init_verbs
/* Macbook 5,1 */
static const struct hda_verb alc885_mb5_init_verbs[] = {
/* DACs */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Front mixer */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Surround mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* LFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* HP mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LFE Pin (0x0e) */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
/* HP Pin (0x0f) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
{ }
};
/* Macmini 3,1 */
static const struct hda_verb alc885_macmini3_init_verbs[] = {
/* DACs */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Front mixer */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Surround mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* LFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* HP mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LFE Pin (0x0e) */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
/* HP Pin (0x0f) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* Line In pin */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{ }
};
static const struct hda_verb alc885_mba21_init_verbs[] = {
/*Internal and HP Speaker Mixer*/
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/*Internal Speaker Pin (0x0c)*/
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* HP Pin: output 0 (0x0e) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
/* Line in (is hp when jack connected)*/
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
/* Macbook Pro rev3 */
static const struct hda_verb alc885_mbp3_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* HP mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* HP Pin: output 0 (0x0e) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: use output 1 when in LineOut mode */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* ADC1: mute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/* iMac 9,1 */
static const struct hda_verb alc885_imac91_init_verbs[] = {
/* Internal Speaker Pin (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
/* HP Pin: Rear */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
/* Line in Rear */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/* Toggle speaker-output according to the hp-jack state */
static void alc885_imac24_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
#define alc885_mb5_setup alc885_imac24_setup
#define alc885_macmini3_setup alc885_imac24_setup
/* Macbook Air 2,1 */
static void alc885_mba21_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc885_mbp3_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc885_imac91_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
static const struct hda_verb alc889A_mb31_ch2_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
{ } /* end */
};
/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
static const struct hda_verb alc889A_mb31_ch4_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
{ } /* end */
};
/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
static const struct hda_verb alc889A_mb31_ch5_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
{ } /* end */
};
/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
static const struct hda_verb alc889A_mb31_ch6_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
{ } /* end */
};
static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
{ 2, alc889A_mb31_ch2_init },
{ 4, alc889A_mb31_ch4_init },
{ 5, alc889A_mb31_ch5_init },
{ 6, alc889A_mb31_ch6_init },
};
static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc889A_mb31_mixer[] = {
/* Output mixers */
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
/* Output switches */
HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
/* Boost mixers */
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
/* Input mixers */
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc883_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static const struct hda_verb alc889A_mb31_verbs[] = {
/* Init rear pin (used as headphone output) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* Init line pin (used as output in 4ch and 6ch mode) */
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */
/* Init line 2 pin (used as headphone out by default) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
{ } /* end */
};
/* Mute speakers according to the headphone jack state */
static void alc889A_mb31_automute(struct hda_codec *codec)
{
unsigned int present;
/* Mute only in 2ch or 4ch mode */
if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
== 0x00) {
present = snd_hda_jack_detect(codec, 0x15);
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
}
static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc889A_mb31_automute(codec);
}
static void alc882_unsol_event(struct hda_codec *codec, unsigned int res)
{
alc_exec_unsol_event(codec, res >> 26);
}
/*
* configuration and preset
*/
static const char * const alc882_models[ALC882_MODEL_LAST] = {
[ALC885_MB5] = "mb5",
[ALC885_MACMINI3] = "macmini3",
[ALC885_MBA21] = "mba21",
[ALC885_MBP3] = "mbp3",
[ALC885_IMAC91] = "imac91",
[ALC889A_MB31] = "mb31",
[ALC882_AUTO] = "auto",
};
/* codec SSID table for Intel Mac */
static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
/* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
* so apparently no perfect solution yet
*/
SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3),
{} /* terminator */
};
static const struct alc_config_preset alc882_presets[] = {
[ALC885_MBA21] = {
.mixers = { alc885_mba21_mixer },
.init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs },
.num_dacs = 2,
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mba21_ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
.input_mux = &alc882_capture_source,
.unsol_event = alc882_unsol_event,
.setup = alc885_mba21_setup,
.init_hook = alc_hp_automute,
},
[ALC885_MBP3] = {
.mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_mbp3_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = 2,
.dac_nids = alc882_dac_nids,
.hp_nid = 0x04,
.channel_mode = alc885_mbp_4ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_mbp3_setup,
.init_hook = alc_hp_automute,
},
[ALC885_MB5] = {
.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_mb5_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mb5_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
.input_mux = &mb5_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_mb5_setup,
.init_hook = alc_hp_automute,
},
[ALC885_MACMINI3] = {
.mixers = { alc885_macmini3_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_macmini3_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_macmini3_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes),
.input_mux = &macmini3_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_macmini3_setup,
.init_hook = alc_hp_automute,
},
[ALC885_IMAC91] = {
.mixers = {alc885_imac91_mixer},
.init_verbs = { alc885_imac91_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mba21_ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
.input_mux = &alc889A_imac91_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_imac91_setup,
.init_hook = alc_hp_automute,
},
[ALC889A_MB31] = {
.mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
.init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
alc880_gpio1_init_verbs },
.adc_nids = alc883_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.capsrc_nids = alc883_capsrc_nids,
.dac_nids = alc883_dac_nids,
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.channel_mode = alc889A_mb31_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
.input_mux = &alc889A_mb31_capture_source,
.dig_out_nid = ALC883_DIGOUT_NID,
.unsol_event = alc889A_mb31_unsol_event,
.init_hook = alc889A_mb31_automute,
},
};

480
sound/pci/hda/alc_quirks.c
View File

@ -1,480 +0,0 @@
/*
* Common codes for Realtek codec quirks
* included by patch_realtek.c
*/
/*
* configuration template - to be copied to the spec instance
*/
struct alc_config_preset {
const struct snd_kcontrol_new *mixers[5]; /* should be identical size
* with spec
*/
const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
const struct hda_verb *init_verbs[5];
unsigned int num_dacs;
const hda_nid_t *dac_nids;
hda_nid_t dig_out_nid; /* optional */
hda_nid_t hp_nid; /* optional */
const hda_nid_t *slave_dig_outs;
unsigned int num_adc_nids;
const hda_nid_t *adc_nids;
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
int need_dac_fix;
int const_channel_count;
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*setup)(struct hda_codec *);
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
const struct hda_amp_list *loopbacks;
void (*power_hook)(struct hda_codec *codec);
#endif
};
/*
* channel mode setting
*/
static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
spec->num_channel_mode);
}
static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
spec->ext_channel_count);
}
static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
&spec->ext_channel_count);
if (err >= 0 && !spec->const_channel_count) {
spec->multiout.max_channels = spec->ext_channel_count;
if (spec->need_dac_fix)
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
}
return err;
}
/*
* Control the mode of pin widget settings via the mixer. "pc" is used
* instead of "%" to avoid consequences of accidentally treating the % as
* being part of a format specifier. Maximum allowed length of a value is
* 63 characters plus NULL terminator.
*
* Note: some retasking pin complexes seem to ignore requests for input
* states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
* are requested. Therefore order this list so that this behaviour will not
* cause problems when mixer clients move through the enum sequentially.
* NIDs 0x0f and 0x10 have been observed to have this behaviour as of
* March 2006.
*/
static const char * const alc_pin_mode_names[] = {
"Mic 50pc bias", "Mic 80pc bias",
"Line in", "Line out", "Headphone out",
};
static const unsigned char alc_pin_mode_values[] = {
PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
};
/* The control can present all 5 options, or it can limit the options based
* in the pin being assumed to be exclusively an input or an output pin. In
* addition, "input" pins may or may not process the mic bias option
* depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
* accept requests for bias as of chip versions up to March 2006) and/or
* wiring in the computer.
*/
#define ALC_PIN_DIR_IN 0x00
#define ALC_PIN_DIR_OUT 0x01
#define ALC_PIN_DIR_INOUT 0x02
#define ALC_PIN_DIR_IN_NOMICBIAS 0x03
#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
/* Info about the pin modes supported by the different pin direction modes.
* For each direction the minimum and maximum values are given.
*/
static const signed char alc_pin_mode_dir_info[5][2] = {
{ 0, 2 }, /* ALC_PIN_DIR_IN */
{ 3, 4 }, /* ALC_PIN_DIR_OUT */
{ 0, 4 }, /* ALC_PIN_DIR_INOUT */
{ 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
{ 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
};
#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
#define alc_pin_mode_n_items(_dir) \
(alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
unsigned int item_num = uinfo->value.enumerated.item;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
item_num = alc_pin_mode_min(dir);
strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
return 0;
}
static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
unsigned int i;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL,
0x00);
/* Find enumerated value for current pinctl setting */
i = alc_pin_mode_min(dir);
while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
i++;
*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
return 0;
}
static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL,
0x00);
if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
val = alc_pin_mode_min(dir);
change = pinctl != alc_pin_mode_values[val];
if (change) {
/* Set pin mode to that requested */
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
alc_pin_mode_values[val]);
/* Also enable the retasking pin's input/output as required
* for the requested pin mode. Enum values of 2 or less are
* input modes.
*
* Dynamically switching the input/output buffers probably
* reduces noise slightly (particularly on input) so we'll
* do it. However, having both input and output buffers
* enabled simultaneously doesn't seem to be problematic if
* this turns out to be necessary in the future.
*/
if (val <= 2) {
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, 0);
} else {
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
}
return change;
}
#define ALC_PIN_MODE(xname, nid, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_pin_mode_info, \
.get = alc_pin_mode_get, \
.put = alc_pin_mode_put, \
.private_value = nid | (dir<<16) }
/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
* together using a mask with more than one bit set. This control is
* currently used only by the ALC260 test model. At this stage they are not
* needed for any "production" models.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_gpio_data_info snd_ctl_boolean_mono_info
static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA,
0x00);
/* Set/unset the masked GPIO bit(s) as needed */
change = (val == 0 ? 0 : mask) != (gpio_data & mask);
if (val == 0)
gpio_data &= ~mask;
else
gpio_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_GPIO_DATA, gpio_data);
return change;
}
#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_gpio_data_info, \
.get = alc_gpio_data_get, \
.put = alc_gpio_data_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling of the digital IO pins on the
* ALC260. This is incredibly simplistic; the intention of this control is
* to provide something in the test model allowing digital outputs to be
* identified if present. If models are found which can utilise these
* outputs a more complete mixer control can be devised for those models if
* necessary.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
if (val==0)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
ctrl_data);
return change;
}
#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_spdif_ctrl_info, \
.get = alc_spdif_ctrl_get, \
.put = alc_spdif_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
* Again, this is only used in the ALC26x test models to help identify when
* the EAPD line must be asserted for features to work.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (!val ? 0 : mask) != (ctrl_data & mask);
if (!val)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
ctrl_data);
return change;
}
#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_eapd_ctrl_info, \
.get = alc_eapd_ctrl_get, \
.put = alc_eapd_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
if (!cfg->line_outs) {
while (cfg->line_outs < AUTO_CFG_MAX_OUTS &&
cfg->line_out_pins[cfg->line_outs])
cfg->line_outs++;
}
if (!cfg->speaker_outs) {
while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS &&
cfg->speaker_pins[cfg->speaker_outs])
cfg->speaker_outs++;
}
if (!cfg->hp_outs) {
while (cfg->hp_outs < AUTO_CFG_MAX_OUTS &&
cfg->hp_pins[cfg->hp_outs])
cfg->hp_outs++;
}
}
/*
* set up from the preset table
*/
static void setup_preset(struct hda_codec *codec,
const struct alc_config_preset *preset)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
add_mixer(spec, preset->mixers[i]);
spec->cap_mixer = preset->cap_mixer;
for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
i++)
add_verb(spec, preset->init_verbs[i]);
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
spec->need_dac_fix = preset->need_dac_fix;
spec->const_channel_count = preset->const_channel_count;
if (preset->const_channel_count)
spec->multiout.max_channels = preset->const_channel_count;
else
spec->multiout.max_channels = spec->channel_mode[0].channels;
spec->ext_channel_count = spec->channel_mode[0].channels;
spec->multiout.num_dacs = preset->num_dacs;
spec->multiout.dac_nids = preset->dac_nids;
spec->multiout.dig_out_nid = preset->dig_out_nid;
spec->multiout.slave_dig_outs = preset->slave_dig_outs;
spec->multiout.hp_nid = preset->hp_nid;
spec->num_mux_defs = preset->num_mux_defs;
if (!spec->num_mux_defs)
spec->num_mux_defs = 1;
spec->input_mux = preset->input_mux;
spec->num_adc_nids = preset->num_adc_nids;
spec->adc_nids = preset->adc_nids;
spec->capsrc_nids = preset->capsrc_nids;
spec->dig_in_nid = preset->dig_in_nid;
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->power_hook = preset->power_hook;
spec->loopback.amplist = preset->loopbacks;
#endif
if (preset->setup)
preset->setup(codec);
alc_fixup_autocfg_pin_nums(codec);
}
static void alc_simple_setup_automute(struct alc_spec *spec, int mode)
{
int lo_pin = spec->autocfg.line_out_pins[0];
if (lo_pin == spec->autocfg.speaker_pins[0] ||
lo_pin == spec->autocfg.hp_pins[0])
lo_pin = 0;
spec->automute_mode = mode;
spec->detect_hp = !!spec->autocfg.hp_pins[0];
spec->detect_lo = !!lo_pin;
spec->automute_lo = spec->automute_lo_possible = !!lo_pin;
spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0];
}
/* auto-toggle front mic */
static void alc88x_simple_mic_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_jack_detect(codec, 0x18);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}

192
sound/pci/hda/hda_codec.c
View File

@ -19,6 +19,7 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <linux/mm.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
@ -2304,7 +2305,7 @@ typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *);
/* apply the function to all matching slave ctls in the mixer list */
static int map_slaves(struct hda_codec *codec, const char * const *slaves,
map_slave_func_t func, void *data)
const char *suffix, map_slave_func_t func, void *data)
{
struct hda_nid_item *items;
const char * const *s;
@ -2317,7 +2318,14 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
continue;
for (s = slaves; *s; s++) {
if (!strcmp(sctl->id.name, *s)) {
char tmpname[sizeof(sctl->id.name)];
const char *name = *s;
if (suffix) {
snprintf(tmpname, sizeof(tmpname), "%s %s",
name, suffix);
name = tmpname;
}
if (!strcmp(sctl->id.name, name)) {
err = func(data, sctl);
if (err)
return err;
@ -2333,12 +2341,65 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
return 1;
}
/* guess the value corresponding to 0dB */
static int get_kctl_0dB_offset(struct snd_kcontrol *kctl)
{
int _tlv[4];
const int *tlv = NULL;
int val = -1;
if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
/* FIXME: set_fs() hack for obtaining user-space TLV data */
mm_segment_t fs = get_fs();
set_fs(get_ds());
if (!kctl->tlv.c(kctl, 0, sizeof(_tlv), _tlv))
tlv = _tlv;
set_fs(fs);
} else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ)
tlv = kctl->tlv.p;
if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE)
val = -tlv[2] / tlv[3];
return val;
}
/* call kctl->put with the given value(s) */
static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
{
struct snd_ctl_elem_value *ucontrol;
ucontrol = kzalloc(sizeof(*ucontrol), GFP_KERNEL);
if (!ucontrol)
return -ENOMEM;
ucontrol->value.integer.value[0] = val;
ucontrol->value.integer.value[1] = val;
kctl->put(kctl, ucontrol);
kfree(ucontrol);
return 0;
}
/* initialize the slave volume with 0dB */
static int init_slave_0dB(void *data, struct snd_kcontrol *slave)
{
int offset = get_kctl_0dB_offset(slave);
if (offset > 0)
put_kctl_with_value(slave, offset);
return 0;
}
/* unmute the slave */
static int init_slave_unmute(void *data, struct snd_kcontrol *slave)
{
return put_kctl_with_value(slave, 1);
}
/**
* snd_hda_add_vmaster - create a virtual master control and add slaves
* @codec: HD-audio codec
* @name: vmaster control name
* @tlv: TLV data (optional)
* @slaves: slave control names (optional)
* @suffix: suffix string to each slave name (optional)
* @init_slave_vol: initialize slaves to unmute/0dB
* @ctl_ret: store the vmaster kcontrol in return
*
* Create a virtual master control with the given name. The TLV data
* must be either NULL or a valid data.
@ -2349,13 +2410,18 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
*
* This function returns zero if successful or a negative error code.
*/
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char * const *slaves)
int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char * const *slaves,
const char *suffix, bool init_slave_vol,
struct snd_kcontrol **ctl_ret)
{
struct snd_kcontrol *kctl;
int err;
err = map_slaves(codec, slaves, check_slave_present, NULL);
if (ctl_ret)
*ctl_ret = NULL;
err = map_slaves(codec, slaves, suffix, check_slave_present, NULL);
if (err != 1) {
snd_printdd("No slave found for %s\n", name);
return 0;
@ -2367,13 +2433,119 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
if (err < 0)
return err;
err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave,
kctl);
err = map_slaves(codec, slaves, suffix,
(map_slave_func_t)snd_ctl_add_slave, kctl);
if (err < 0)
return err;
/* init with master mute & zero volume */
put_kctl_with_value(kctl, 0);
if (init_slave_vol)
map_slaves(codec, slaves, suffix,
tlv ? init_slave_0dB : init_slave_unmute, kctl);
if (ctl_ret)
*ctl_ret = kctl;
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_add_vmaster);
EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster);
/*
* mute-LED control using vmaster
*/
static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char * const texts[] = {
"Off", "On", "Follow Master"
};
unsigned int index;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 3;
index = uinfo->value.enumerated.item;
if (index >= 3)
index = 2;
strcpy(uinfo->value.enumerated.name, texts[index]);
return 0;
}
static int vmaster_mute_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = hook->mute_mode;
return 0;
}
static int vmaster_mute_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol);
unsigned int old_mode = hook->mute_mode;
hook->mute_mode = ucontrol->value.enumerated.item[0];
if (hook->mute_mode > HDA_VMUTE_FOLLOW_MASTER)
hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER;
if (old_mode == hook->mute_mode)
return 0;
snd_hda_sync_vmaster_hook(hook);
return 1;
}
static struct snd_kcontrol_new vmaster_mute_mode = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Mute-LED Mode",
.info = vmaster_mute_mode_info,
.get = vmaster_mute_mode_get,
.put = vmaster_mute_mode_put,
};
/*
* Add a mute-LED hook with the given vmaster switch kctl
* "Mute-LED Mode" control is automatically created and associated with
* the given hook.
*/
int snd_hda_add_vmaster_hook(struct hda_codec *codec,
struct hda_vmaster_mute_hook *hook,
bool expose_enum_ctl)
{
struct snd_kcontrol *kctl;
if (!hook->hook || !hook->sw_kctl)
return 0;
snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec);
hook->codec = codec;
hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER;
if (!expose_enum_ctl)
return 0;
kctl = snd_ctl_new1(&vmaster_mute_mode, hook);
if (!kctl)
return -ENOMEM;
return snd_hda_ctl_add(codec, 0, kctl);
}
EXPORT_SYMBOL_HDA(snd_hda_add_vmaster_hook);
/*
* Call the hook with the current value for synchronization
* Should be called in init callback
*/
void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook)
{
if (!hook->hook || !hook->codec)
return;
switch (hook->mute_mode) {
case HDA_VMUTE_FOLLOW_MASTER:
snd_ctl_sync_vmaster_hook(hook->sw_kctl);
break;
default:
hook->hook(hook->codec, hook->mute_mode);
break;
}
}
EXPORT_SYMBOL_HDA(snd_hda_sync_vmaster_hook);
/**
* snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch
@ -5272,6 +5444,10 @@ int snd_hda_suspend(struct hda_bus *bus)
list_for_each_entry(codec, &bus->codec_list, list) {
if (hda_codec_is_power_on(codec))
hda_call_codec_suspend(codec);
else /* forcibly change the power to D3 even if not used */
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D3);
if (codec->patch_ops.post_suspend)
codec->patch_ops.post_suspend(codec);
}

1
sound/pci/hda/hda_codec.h
View File

@ -855,6 +855,7 @@ struct hda_codec {
unsigned int pins_shutup:1; /* pins are shut up */
unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
unsigned int no_jack_detect:1; /* Machine has no jack-detection */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
unsigned int power_transition :1; /* power-state in transition */

4
sound/pci/hda/hda_eld.c
View File

@ -385,8 +385,8 @@ error:
static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen)
{
static unsigned int alsa_rates[] = {
5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
96000, 176400, 192000, 384000
5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
88200, 96000, 176400, 192000, 384000
};
int i, j;

52
sound/pci/hda/hda_intel.c
View File

@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444);
MODULE_PARM_DESC(model, "Use the given board model.");
module_param_array(position_fix, int, NULL, 0444);
MODULE_PARM_DESC(position_fix, "DMA pointer read method."
"(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO).");
"(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO).");
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);
@ -94,7 +94,7 @@ 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).");
module_param(enable_msi, int, 0444);
module_param(enable_msi, bint, 0444);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
#ifdef CONFIG_SND_HDA_PATCH_LOADER
module_param_array(patch, charp, NULL, 0444);
@ -121,8 +121,8 @@ module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif
static bool align_buffer_size = 1;
module_param(align_buffer_size, bool, 0644);
static int align_buffer_size = -1;
module_param(align_buffer_size, bint, 0644);
MODULE_PARM_DESC(align_buffer_size,
"Force buffer and period sizes to be multiple of 128 bytes.");
@ -148,6 +148,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, PCH},"
"{Intel, CPT},"
"{Intel, PPT},"
"{Intel, LPT},"
"{Intel, PBG},"
"{Intel, SCH},"
"{ATI, SB450},"
@ -329,6 +330,7 @@ enum {
POS_FIX_LPIB,
POS_FIX_POSBUF,
POS_FIX_VIACOMBO,
POS_FIX_COMBO,
};
/* Defines for ATI HD Audio support in SB450 south bridge */
@ -515,6 +517,7 @@ enum {
#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */
#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@ -527,7 +530,8 @@ enum {
/* quirks for Nvidia */
#define AZX_DCAPS_PRESET_NVIDIA \
(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI)
(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
AZX_DCAPS_ALIGN_BUFSIZE)
static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_ICH] = "HDA Intel",
@ -2347,17 +2351,6 @@ static void azx_power_notify(struct hda_bus *bus)
* 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);
@ -2404,8 +2397,7 @@ static int azx_resume(struct pci_dev *pci)
return -EIO;
azx_init_pci(chip);
if (snd_hda_codecs_inuse(chip->bus))
azx_init_chip(chip, 1);
azx_init_chip(chip, 1);
snd_hda_resume(chip->bus);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
@ -2517,6 +2509,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
case POS_FIX_LPIB:
case POS_FIX_POSBUF:
case POS_FIX_VIACOMBO:
case POS_FIX_COMBO:
return fix;
}
@ -2696,6 +2689,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->position_fix[0] = chip->position_fix[1] =
check_position_fix(chip, position_fix[dev]);
/* combo mode uses LPIB for playback */
if (chip->position_fix[0] == POS_FIX_COMBO) {
chip->position_fix[0] = POS_FIX_LPIB;
chip->position_fix[1] = POS_FIX_AUTO;
}
check_probe_mask(chip, dev);
chip->single_cmd = single_cmd;
@ -2774,9 +2773,16 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
/* disable buffer size rounding to 128-byte multiples if supported */
chip->align_buffer_size = align_buffer_size;
if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
chip->align_buffer_size = 0;
if (align_buffer_size >= 0)
chip->align_buffer_size = !!align_buffer_size;
else {
if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
chip->align_buffer_size = 0;
else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE)
chip->align_buffer_size = 1;
else
chip->align_buffer_size = 1;
}
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
@ -2992,6 +2998,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ PCI_DEVICE(0x8086, 0x1e20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE},
/* Lynx Point */
{ PCI_DEVICE(0x8086, 0x8c20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE},
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |

16
sound/pci/hda/hda_jack.c
View File

@ -19,6 +19,22 @@
#include "hda_local.h"
#include "hda_jack.h"
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
{
if (codec->no_jack_detect)
return false;
if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
return false;
if (!codec->ignore_misc_bit &&
(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
AC_DEFCFG_MISC_NO_PRESENCE))
return false;
if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
return false;
return true;
}
EXPORT_SYMBOL_HDA(is_jack_detectable);
/* execute pin sense measurement */
static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
{

13
sound/pci/hda/hda_jack.h
View File

@ -62,18 +62,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
{
if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
return false;
if (!codec->ignore_misc_bit &&
(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
AC_DEFCFG_MISC_NO_PRESENCE))
return false;
if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
return false;
return true;
}
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
const char *name, int idx);

30
sound/pci/hda/hda_local.h
View File

@ -139,10 +139,36 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int *tlv);
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 * const *slaves);
int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char * const *slaves,
const char *suffix, bool init_slave_vol,
struct snd_kcontrol **ctl_ret);
#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \
__snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL)
int snd_hda_codec_reset(struct hda_codec *codec);
enum {
HDA_VMUTE_OFF,
HDA_VMUTE_ON,
HDA_VMUTE_FOLLOW_MASTER,
};
struct hda_vmaster_mute_hook {
/* below two fields must be filled by the caller of
* snd_hda_add_vmaster_hook() beforehand
*/
struct snd_kcontrol *sw_kctl;
void (*hook)(void *, int);
/* below are initialized automatically */
unsigned int mute_mode; /* HDA_VMUTE_XXX */
struct hda_codec *codec;
};
int snd_hda_add_vmaster_hook(struct hda_codec *codec,
struct hda_vmaster_mute_hook *hook,
bool expose_enum_ctl);
void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook);
/* amp value bits */
#define HDA_AMP_MUTE 0x80
#define HDA_AMP_UNMUTE 0x00

72
sound/pci/hda/patch_analog.c
View File

@ -82,6 +82,7 @@ struct ad198x_spec {
unsigned int inv_jack_detect: 1;/* inverted jack-detection */
unsigned int inv_eapd: 1; /* inverted EAPD implementation */
unsigned int analog_beep: 1; /* analog beep input present */
unsigned int avoid_init_slave_vol:1;
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
@ -137,51 +138,17 @@ static int ad198x_init(struct hda_codec *codec)
return 0;
}
static const char * const ad_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Mono Playback Volume",
"Speaker Playback Volume",
"IEC958 Playback Volume",
static const char * const ad_slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
"Headphone", "Mono", "Speaker", "IEC958",
NULL
};
static const char * const ad_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Mono Playback Switch",
"Speaker Playback Switch",
"IEC958 Playback Switch",
static const char * const ad1988_6stack_fp_slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side", "IEC958",
NULL
};
static const char * const ad1988_6stack_fp_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"IEC958 Playback Volume",
NULL
};
static const char * const ad1988_6stack_fp_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"IEC958 Playback Switch",
NULL
};
static void ad198x_free_kctls(struct hda_codec *codec);
#ifdef CONFIG_SND_HDA_INPUT_BEEP
@ -257,10 +224,12 @@ static int ad198x_build_controls(struct hda_codec *codec)
unsigned int vmaster_tlv[4];
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv,
(spec->slave_vols ?
spec->slave_vols : ad_slave_vols));
spec->slave_vols : ad_slave_pfxs),
"Playback Volume",
!spec->avoid_init_slave_vol, NULL);
if (err < 0)
return err;
}
@ -268,7 +237,8 @@ static int ad198x_build_controls(struct hda_codec *codec)
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL,
(spec->slave_sws ?
spec->slave_sws : ad_slave_sws));
spec->slave_sws : ad_slave_pfxs),
"Playback Switch");
if (err < 0)
return err;
}
@ -3385,8 +3355,8 @@ static int patch_ad1988(struct hda_codec *codec)
if (spec->autocfg.hp_pins[0]) {
spec->mixers[spec->num_mixers++] = ad1988_hp_mixers;
spec->slave_vols = ad1988_6stack_fp_slave_vols;
spec->slave_sws = ad1988_6stack_fp_slave_sws;
spec->slave_vols = ad1988_6stack_fp_slave_pfxs;
spec->slave_sws = ad1988_6stack_fp_slave_pfxs;
spec->alt_dac_nid = ad1988_alt_dac_nid;
spec->stream_analog_alt_playback =
&ad198x_pcm_analog_alt_playback;
@ -3594,16 +3564,8 @@ static const struct hda_amp_list ad1884_loopbacks[] = {
#endif
static const char * const ad1884_slave_vols[] = {
"PCM Playback Volume",
"Mic Playback Volume",
"Mono Playback Volume",
"Front Mic Playback Volume",
"Mic Playback Volume",
"CD Playback Volume",
"Internal Mic Playback Volume",
"Docking Mic Playback Volume",
/* "Beep Playback Volume", */
"IEC958 Playback Volume",
"PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
"Internal Mic", "Docking Mic", /* "Beep", */ "IEC958",
NULL
};
@ -3644,6 +3606,8 @@ static int patch_ad1884(struct hda_codec *codec)
spec->vmaster_nid = 0x04;
/* we need to cover all playback volumes */
spec->slave_vols = ad1884_slave_vols;
/* slaves may contain input volumes, so we can't raise to 0dB blindly */
spec->avoid_init_slave_vol = 1;
codec->patch_ops = ad198x_patch_ops;

122
sound/pci/hda/patch_conexant.c
View File

@ -70,6 +70,8 @@ struct conexant_spec {
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
hda_nid_t vmaster_nid;
struct hda_vmaster_mute_hook vmaster_mute;
bool vmaster_mute_led;
const struct hda_verb *init_verbs[5]; /* initialization verbs
* don't forget NULL
@ -465,21 +467,8 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = {
};
#endif
static const char * const slave_vols[] = {
"Headphone Playback Volume",
"Speaker Playback Volume",
"Front Playback Volume",
"Surround Playback Volume",
"CLFE Playback Volume",
NULL
};
static const char * const slave_sws[] = {
"Headphone Playback Switch",
"Speaker Playback Switch",
"Front Playback Switch",
"Surround Playback Switch",
"CLFE Playback Switch",
static const char * const slave_pfxs[] = {
"Headphone", "Speaker", "Front", "Surround", "CLFE",
NULL
};
@ -519,14 +508,17 @@ static int conexant_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, slave_vols);
vmaster_tlv, slave_pfxs,
"Playback Volume");
if (err < 0)
return err;
}
if (spec->vmaster_nid &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_sws);
err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_pfxs,
"Playback Switch", true,
&spec->vmaster_mute.sw_kctl);
if (err < 0)
return err;
}
@ -3034,7 +3026,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
{}
};
@ -3943,6 +3934,63 @@ static void enable_unsol_pins(struct hda_codec *codec, int num_pins,
snd_hda_jack_detect_enable(codec, pins[i], action);
}
static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
{
int i;
for (i = 0; i < nums; i++)
if (list[i] == nid)
return true;
return false;
}
/* is the given NID found in any of autocfg items? */
static bool found_in_autocfg(struct auto_pin_cfg *cfg, hda_nid_t nid)
{
int i;
if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) ||
found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) ||
found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs) ||
found_in_nid_list(nid, cfg->dig_out_pins, cfg->dig_outs))
return true;
for (i = 0; i < cfg->num_inputs; i++)
if (cfg->inputs[i].pin == nid)
return true;
if (cfg->dig_in_pin == nid)
return true;
return false;
}
/* clear unsol-event tags on unused pins; Conexant codecs seem to leave
* invalid unsol tags by some reason
*/
static void clear_unsol_on_unused_pins(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
int i;
for (i = 0; i < codec->init_pins.used; i++) {
struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
if (!found_in_autocfg(cfg, pin->nid))
snd_hda_codec_write(codec, pin->nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE, 0);
}
}
/* turn on/off EAPD according to Master switch */
static void cx_auto_vmaster_hook(void *private_data, int enabled)
{
struct hda_codec *codec = private_data;
struct conexant_spec *spec = codec->spec;
if (enabled && spec->pin_eapd_ctrls) {
cx_auto_update_speakers(codec);
return;
}
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, enabled);
}
static void cx_auto_init_output(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@ -3983,6 +4031,7 @@ static void cx_auto_init_output(struct hda_codec *codec)
/* turn on all EAPDs if no individual EAPD control is available */
if (!spec->pin_eapd_ctrls)
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
clear_unsol_on_unused_pins(codec);
}
static void cx_auto_init_input(struct hda_codec *codec)
@ -4046,11 +4095,13 @@ static void cx_auto_init_digital(struct hda_codec *codec)
static int cx_auto_init(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
/*snd_hda_sequence_write(codec, cx_auto_init_verbs);*/
cx_auto_init_output(codec);
cx_auto_init_input(codec);
cx_auto_init_digital(codec);
snd_hda_jack_report_sync(codec);
snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
return 0;
}
@ -4296,6 +4347,13 @@ static int cx_auto_build_controls(struct hda_codec *codec)
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
if (err < 0)
return err;
if (spec->vmaster_mute.sw_kctl) {
spec->vmaster_mute.hook = cx_auto_vmaster_hook;
err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute,
spec->vmaster_mute_led);
if (err < 0)
return err;
}
return 0;
}
@ -4320,7 +4378,6 @@ static int cx_auto_search_adcs(struct hda_codec *codec)
return 0;
}
static const struct hda_codec_ops cx_auto_patch_ops = {
.build_controls = cx_auto_build_controls,
.build_pcms = conexant_build_pcms,
@ -4368,6 +4425,7 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
{ 0x17, 0x21a11000 }, /* dock-mic */
{ 0x19, 0x2121103f }, /* dock-HP */
{ 0x1c, 0x21440100 }, /* dock SPDIF out */
{}
};
@ -4421,6 +4479,18 @@ static int patch_conexant_auto(struct hda_codec *codec)
apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
/* Show mute-led control only on HP laptops
* This is a sort of white-list: on HP laptops, EAPD corresponds
* only to the mute-LED without actualy amp function. Meanwhile,
* others may use EAPD really as an amp switch, so it might be
* not good to expose it blindly.
*/
switch (codec->subsystem_id >> 16) {
case 0x103c:
spec->vmaster_mute_led = 1;
break;
}
err = cx_auto_search_adcs(codec);
if (err < 0)
return err;
@ -4434,6 +4504,18 @@ static int patch_conexant_auto(struct hda_codec *codec)
codec->patch_ops = cx_auto_patch_ops;
if (spec->beep_amp)
snd_hda_attach_beep_device(codec, spec->beep_amp);
/* Some laptops with Conexant chips show stalls in S3 resume,
* which falls into the single-cmd mode.
* Better to make reset, then.
*/
if (!codec->bus->sync_write) {
snd_printd("hda_codec: "
"Enable sync_write for stable communication\n");
codec->bus->sync_write = 1;
codec->bus->allow_bus_reset = 1;
}
return 0;
}

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

@ -1912,6 +1912,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi },
{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi },
{} /* terminator */
};
@ -1958,6 +1959,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862803");
MODULE_ALIAS("snd-hda-codec-id:80862804");
MODULE_ALIAS("snd-hda-codec-id:80862805");
MODULE_ALIAS("snd-hda-codec-id:80862806");
MODULE_ALIAS("snd-hda-codec-id:80862880");
MODULE_ALIAS("snd-hda-codec-id:808629fb");
MODULE_LICENSE("GPL");

1930
sound/pci/hda/patch_realtek.c
View File

File diff suppressed because it is too large Load Diff

203
sound/pci/hda/patch_sigmatel.c
View File

@ -99,6 +99,7 @@ enum {
STAC_DELL_VOSTRO_3500,
STAC_92HD83XXX_HP_cNB11_INTQUAD,
STAC_HP_DV7_4000,
STAC_HP_ZEPHYR,
STAC_92HD83XXX_MODELS
};
@ -309,6 +310,8 @@ struct sigmatel_spec {
unsigned long auto_capvols[MAX_ADCS_NUM];
unsigned auto_dmic_cnt;
hda_nid_t auto_dmic_nids[MAX_DMICS_NUM];
struct hda_vmaster_mute_hook vmaster_mute;
};
static const hda_nid_t stac9200_adc_nids[1] = {
@ -662,7 +665,6 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
return 0;
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac_vrefout_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
{
@ -686,7 +688,6 @@ static int stac_vrefout_set(struct hda_codec *codec,
return 1;
}
#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@ -894,6 +895,13 @@ static const struct hda_verb stac92hd83xxx_core_init[] = {
{}
};
static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = {
{ 0x22, 0x785, 0x43 },
{ 0x22, 0x782, 0xe0 },
{ 0x22, 0x795, 0x00 },
{}
};
static const struct hda_verb stac92hd71bxx_core_init[] = {
/* set master volume and direct control */
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@ -999,8 +1007,8 @@ static const struct hda_verb stac9205_core_init[] = {
}
static const struct snd_kcontrol_new stac9200_mixer[] = {
HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
{ } /* end */
@ -1027,8 +1035,8 @@ static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
};
static const struct snd_kcontrol_new stac925x_mixer[] = {
HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xe, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x0e, 0, HDA_OUTPUT),
{ } /* end */
};
@ -1060,34 +1068,25 @@ static struct snd_kcontrol_new stac_smux_mixer = {
.put = stac92xx_smux_enum_put,
};
static const char * const slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Speaker Playback Volume",
static const char * const slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
"Headphone", "Speaker", "IEC958",
NULL
};
static const char * const slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Speaker Playback Switch",
"IEC958 Playback Switch",
NULL
};
static void stac92xx_update_led_status(struct hda_codec *codec, int enabled);
static void stac92xx_vmaster_hook(void *private_data, int val)
{
stac92xx_update_led_status(private_data, val);
}
static void stac92xx_free_kctls(struct hda_codec *codec);
static int stac92xx_build_controls(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
unsigned int vmaster_tlv[4];
int err;
int i;
@ -1144,22 +1143,28 @@ static int stac92xx_build_controls(struct hda_codec *codec)
}
/* if we have no master control, let's create it */
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
unsigned int vmaster_tlv[4];
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
/* correct volume offset */
vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
/* minimum value is actually mute */
vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, slave_vols);
if (err < 0)
return err;
}
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_sws);
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
/* correct volume offset */
vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
/* minimum value is actually mute */
vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, slave_pfxs,
"Playback Volume");
if (err < 0)
return err;
err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_pfxs,
"Playback Switch", true,
&spec->vmaster_mute.sw_kctl);
if (err < 0)
return err;
if (spec->gpio_led) {
spec->vmaster_mute.hook = stac92xx_vmaster_hook;
err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true);
if (err < 0)
return err;
}
@ -1636,6 +1641,12 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = {
0x40f000f0, 0x40f000f0,
};
static const unsigned int hp_zephyr_pin_configs[10] = {
0x01813050, 0x0421201f, 0x04a1205e, 0x96130310,
0x96130310, 0x0101401f, 0x1111611f, 0xd5a30130,
0, 0,
};
static const unsigned int hp_cNB11_intquad_pin_configs[10] = {
0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110,
0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130,
@ -1649,6 +1660,7 @@ static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
[STAC_DELL_VOSTRO_3500] = dell_vostro_3500_pin_configs,
[STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs,
[STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs,
[STAC_HP_ZEPHYR] = hp_zephyr_pin_configs,
};
static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
@ -1659,6 +1671,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
[STAC_DELL_VOSTRO_3500] = "dell-vostro-3500",
[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
[STAC_HP_DV7_4000] = "hp-dv7-4000",
[STAC_HP_ZEPHYR] = "hp-zephyr",
};
static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@ -1711,6 +1724,14 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
"HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593,
"HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
"HP", STAC_HP_ZEPHYR),
{} /* terminator */
};
static const struct snd_pci_quirk stac92hd83xxx_codec_id_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
"HP", STAC_HP_ZEPHYR),
{} /* terminator */
};
@ -4410,8 +4431,7 @@ static int stac92xx_init(struct hda_codec *codec)
snd_hda_jack_report_sync(codec);
/* sync mute LED */
if (spec->gpio_led)
hda_call_check_power_status(codec, 0x01);
snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
if (spec->dac_list)
stac92xx_power_down(codec);
return 0;
@ -4989,7 +5009,6 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
return 0;
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac92xx_pre_resume(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@ -5024,83 +5043,41 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
afg_power_state);
snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
}
#else
#define stac92xx_suspend NULL
#define stac92xx_resume NULL
#define stac92xx_pre_resume NULL
#define stac92xx_set_power_state NULL
#endif /* CONFIG_PM */
/*
* For this feature CONFIG_SND_HDA_POWER_SAVE is needed
* as mute LED state is updated in check_power_status hook
*/
static int stac92xx_update_led_status(struct hda_codec *codec)
/* update mute-LED accoring to the master switch */
static void stac92xx_update_led_status(struct hda_codec *codec, int enabled)
{
struct sigmatel_spec *spec = codec->spec;
int i, num_ext_dacs, muted = 1;
unsigned int muted_lvl, notmtd_lvl;
hda_nid_t nid;
int muted = !enabled;
if (!spec->gpio_led)
return 0;
return;
/* LED state is inverted on these systems */
if (spec->gpio_led_polarity)
muted = !muted;
for (i = 0; i < spec->multiout.num_dacs; i++) {
nid = spec->multiout.dac_nids[i];
if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
HDA_AMP_MUTE)) {
muted = 0; /* something heard */
break;
}
}
if (muted && spec->multiout.hp_nid)
if (!(snd_hda_codec_amp_read(codec,
spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) &
HDA_AMP_MUTE)) {
muted = 0; /* HP is not muted */
}
num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid);
for (i = 0; muted && i < num_ext_dacs; i++) {
nid = spec->multiout.extra_out_nid[i];
if (nid == 0)
break;
if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
HDA_AMP_MUTE)) {
muted = 0; /* extra output is not muted */
}
}
/*polarity defines *not* muted state level*/
if (!spec->vref_mute_led_nid) {
if (muted)
spec->gpio_data &= ~spec->gpio_led; /* orange */
else
spec->gpio_data |= spec->gpio_led; /* white */
if (!spec->gpio_led_polarity) {
/* LED state is inverted on these systems */
spec->gpio_data ^= spec->gpio_led;
}
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir, spec->gpio_data);
} else {
notmtd_lvl = spec->gpio_led_polarity ?
AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
muted_lvl = spec->gpio_led_polarity ?
AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50;
spec->vref_led = muted ? muted_lvl : notmtd_lvl;
spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
stac_vrefout_set(codec, spec->vref_mute_led_nid,
spec->vref_led);
}
return 0;
}
/*
* use power check for controlling mute led of HP notebooks
*/
static int stac92xx_check_power_status(struct hda_codec *codec,
hda_nid_t nid)
{
stac92xx_update_led_status(codec);
return 0;
}
#endif /* CONFIG_SND_HDA_POWER_SAVE */
#endif /* CONFIG_PM */
static const struct hda_codec_ops stac92xx_patch_ops = {
.build_controls = stac92xx_build_controls,
.build_pcms = stac92xx_build_pcms,
@ -5580,6 +5557,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
STAC_92HD83XXX_MODELS,
stac92hd83xxx_models,
stac92hd83xxx_cfg_tbl);
/* check codec subsystem id if not found */
if (spec->board_config < 0)
spec->board_config =
snd_hda_check_board_codec_sid_config(codec,
STAC_92HD83XXX_MODELS, stac92hd83xxx_models,
stac92hd83xxx_codec_id_cfg_tbl);
again:
if (spec->board_config < 0)
snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
@ -5590,12 +5573,17 @@ again:
codec->patch_ops = stac92xx_patch_ops;
switch (spec->board_config) {
case STAC_HP_ZEPHYR:
spec->init = stac92hd83xxx_hp_zephyr_init;
break;
}
if (find_mute_led_cfg(codec, -1/*no default cfg*/))
snd_printd("mute LED gpio %d polarity %d\n",
spec->gpio_led,
spec->gpio_led_polarity);
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
@ -5605,11 +5593,10 @@ again:
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
#ifdef CONFIG_PM
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
stac92xx_check_power_status;
#endif
}
#endif
err = stac92xx_parse_auto_config(codec);
if (!err) {
@ -5906,7 +5893,6 @@ again:
spec->gpio_led,
spec->gpio_led_polarity);
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
@ -5916,11 +5902,10 @@ again:
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
#ifdef CONFIG_PM
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
stac92xx_check_power_status;
#endif
}
#endif
spec->multiout.dac_nids = spec->dac_nids;

48
sound/pci/hda/patch_via.c
View File

@ -550,7 +550,10 @@ static void via_auto_init_output(struct hda_codec *codec,
pin = path->path[path->depth - 1];
init_output_pin(codec, pin, pin_type);
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
else
caps = 0;
if (caps & AC_AMPCAP_MUTE) {
unsigned int val;
val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
@ -645,6 +648,10 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
/* init ADCs */
for (i = 0; i < spec->num_adc_nids; i++) {
hda_nid_t nid = spec->adc_nids[i];
if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP) ||
!(query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE))
continue;
snd_hda_codec_write(codec, spec->adc_nids[i], 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
@ -1445,25 +1452,9 @@ static const struct hda_pcm_stream via_pcm_digital_capture = {
/*
* slave controls for virtual master
*/
static const char * const via_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Speaker Playback Volume",
NULL,
};
static const char * const via_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Speaker Playback Switch",
static const char * const via_slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
"Headphone", "Speaker",
NULL,
};
@ -1508,13 +1499,15 @@ static int via_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, via_slave_vols);
vmaster_tlv, via_slave_pfxs,
"Playback Volume");
if (err < 0)
return err;
}
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, via_slave_sws);
NULL, via_slave_pfxs,
"Playback Switch");
if (err < 0)
return err;
}
@ -1522,6 +1515,8 @@ static int via_build_controls(struct hda_codec *codec)
/* assign Capture Source enums to NID */
kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
for (i = 0; kctl && i < kctl->count; i++) {
if (!spec->mux_nids[i])
continue;
err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]);
if (err < 0)
return err;
@ -2488,6 +2483,8 @@ static int create_mic_boost_ctls(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
const struct auto_pin_cfg *cfg = &spec->autocfg;
const char *prev_label = NULL;
int type_idx = 0;
int i, err;
for (i = 0; i < cfg->num_inputs; i++) {
@ -2502,8 +2499,13 @@ static int create_mic_boost_ctls(struct hda_codec *codec)
if (caps == -1 || !(caps & AC_AMPCAP_NUM_STEPS))
continue;
label = hda_get_autocfg_input_label(codec, cfg, i);
if (prev_label && !strcmp(label, prev_label))
type_idx++;
else
type_idx = 0;
prev_label = label;
snprintf(name, sizeof(name), "%s Boost Volume", label);
err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
err = __via_add_control(spec, VIA_CTL_WIDGET_VOL, name, type_idx,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT));
if (err < 0)
return err;

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

@ -1013,6 +1013,25 @@ static int set_rate_constraints(struct snd_ice1712 *ice,
ice->hw_rates);
}
/* if the card has the internal rate locked (is_pro_locked), limit runtime
hw rates to the current internal rate only.
*/
static void constrain_rate_if_locked(struct snd_pcm_substream *substream)
{
struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned int rate;
if (is_pro_rate_locked(ice)) {
rate = ice->get_rate(ice);
if (rate >= runtime->hw.rate_min
&& rate <= runtime->hw.rate_max) {
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
}
}
/* multi-channel playback needs alignment 8x32bit regardless of the channels
* actually used
*/
@ -1046,6 +1065,7 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
constrain_rate_if_locked(substream);
if (ice->pro_open)
ice->pro_open(ice, substream);
return 0;
@ -1066,6 +1086,7 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
constrain_rate_if_locked(substream);
if (ice->pro_open)
ice->pro_open(ice, substream);
return 0;
@ -1215,6 +1236,7 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
constrain_rate_if_locked(substream);
if (ice->spdif.ops.open)
ice->spdif.ops.open(ice, substream);
return 0;
@ -1251,6 +1273,7 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
constrain_rate_if_locked(substream);
if (ice->spdif.ops.open)
ice->spdif.ops.open(ice, substream);
return 0;

9
sound/pci/ymfpci/ymfpci_main.c
View File

@ -2317,6 +2317,10 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state)
for (i = 0; i < YDSXGR_NUM_SAVED_REGS; i++)
chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]);
chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE);
pci_read_config_word(chip->pci, PCIR_DSXG_LEGACY,
&chip->saved_dsxg_legacy);
pci_read_config_word(chip->pci, PCIR_DSXG_ELEGACY,
&chip->saved_dsxg_elegacy);
snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0);
snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0);
snd_ymfpci_disable_dsp(chip);
@ -2351,6 +2355,11 @@ int snd_ymfpci_resume(struct pci_dev *pci)
snd_ac97_resume(chip->ac97);
pci_write_config_word(chip->pci, PCIR_DSXG_LEGACY,
chip->saved_dsxg_legacy);
pci_write_config_word(chip->pci, PCIR_DSXG_ELEGACY,
chip->saved_dsxg_elegacy);
/* start hw again */
if (chip->start_count > 0) {
spin_lock_irq(&chip->reg_lock);

3
sound/soc/Kconfig
View File

@ -25,6 +25,9 @@ if SND_SOC
config SND_SOC_AC97_BUS
bool
config SND_SOC_DMAENGINE_PCM
bool
# All the supported SoCs
source "sound/soc/atmel/Kconfig"
source "sound/soc/au1x/Kconfig"

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