OpenE2K
/
linux
6
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

sound updates for 4.9-rc1 

Again the diffstat shows a widely distributed pattern at this cycle,
 as there've been many code cleanups and refactoring allover the places.
 Other than that, the development was relatively calm, and no big
 surprise shouldn't be expected.  Here are some highlights:
 
 Core:
 - Sequencer code refactoring / documentation updates
 - TLV code moved to uapi, following some relevant cleanups
 
 USB-Audio:
 - Lots of LINE6 driver fixes / updates
 - DragonFly and TEAC device quirk updates
 
 HD-audio:
 - Usual fixupes for Dell, Lenovo and HP machines
 - Link-audio time reporting capability
 
 ASoC:
 - Large refactoring of simple-card code to be shared with rcar
   driver
 - Removal of some duplicated ops over lots of CODEC drivers
 - Again quite a few Intel SKL driver updates
 - New drivers for Nuvoton NAU88C10, Realtek RT5660 and RT5663
 -----BEGIN PGP SIGNATURE-----
 Version: GnuPG v2.0.22 (GNU/Linux)
 
 iQIcBAABAgAGBQJX9UEbAAoJEGwxgFQ9KSmkq9wQAJ05VA7gEdx0H8vce+zefLDM
 cgFgj6WxMNmfTkPLDpcgUlk2p3GVkJRv1Ny92WlkdE+WBMiooO6qrDUd9ODcq4P4
 ChzJ0AUS33UdzLgd1HbTaRDfea6/49zndlMP6SBDQxB6SUEtklcekgfcFstDnuk6
 l0aFuxOuqnuIVg05UmDGi9lQYk+G18HQYjk6kCLk76l/dzVSaxDV5Tvi/z/DCeIz
 GRSgnjtdmyHZdoQcaBLmLtUr65LRZKWcXgRxATXGdZnZJ2Kg1RZ/aIcbUxdZlFG4
 xHPChWX4MWKOwNJrN1AdRRFEIP1KezSUUzu1yIpjIYpQicuWJO11KaKsCtLwgUui
 g3V8+1niah6zT8zDy5ijpca0pPJg0CPjvEDIOTiy+47awUmO0pfR8kYuT3xWaD6p
 7cHRwiRFI9f0xI5wrHg3Dncw/tGJZ5dZvM3lP0P3qh/7H7bxFW5JmXXUI/Jl2lGo
 FyWgOYnDLF+vPQTcOsupkVnxW9aqSd29smKtfilB9DRXu5i8FT1iM7YllPVS+jRs
 RTdR3TfrF/FjoxWCuVxH5L8YNbZLlxSCVl2+8Kh6IAIDUn2jU138D3iKfx/n4xE8
 sBelEUoSQrFta0AlKQwMerObWIdpTmVkoVtQ5E1lO0fhHgUBZGaQXY7BAv60yjdm
 YoWf2lPBzyUAUf0K1sxd
 =LbA0
 -----END PGP SIGNATURE-----

Merge tag 'sound-4.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Again the diffstat shows a widely distributed pattern at this cycle,
  as there've been many code cleanups and refactoring allover the
  places. Other than that, the development was relatively calm, and no
  big surprise shouldn't be expected. Here are some highlights:

  Core:
   - Sequencer code refactoring / documentation updates
   - TLV code moved to uapi, following some relevant cleanups

  USB-Audio:
   - Lots of LINE6 driver fixes / updates
   - DragonFly and TEAC device quirk updates

  HD-audio:
   - Usual fixupes for Dell, Lenovo and HP machines
   - Link-audio time reporting capability

  ASoC:
   - Large refactoring of simple-card code to be shared with rcar driver
   - Removal of some duplicated ops over lots of CODEC drivers
   - Again quite a few Intel SKL driver updates
   - New drivers for Nuvoton NAU88C10, Realtek RT5660 and RT5663"

* tag 'sound-4.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (465 commits)
  ASoC: fsl: Fix lockups with recent cache changes
  ASoC: Intel: Skylake: fix memory leak of module on error exit path
  ASoC: rsnd: add SNDRV_PCM_TRIGGER_SUSPEND/RESUME
  ASoC: wm8960: remove usage of obsoleted TLV-related macro
  ASoC: rt5616: remove usage of obsoleted TLV-related macro
  ASoC: max9867: remove usage of obsoleted TLV-related macro
  ASoC: trivial: system spelling fix
  ASoC: da7219: fix inappropriate condition statement
  ASoC: tlv320aic31xx: do not declare support for mono DAI
  ASoC: stac9766: fix wrong usage of DECLARE_TLV_DB_LINEAR()
  ASoC: wm8991: remove unused variable
  ASoC: wm8991: fix wrong usage of DECLARE_TLV_DB_LINEAR()
  ASOC: tpa6130a2: add static qualifier for file local symbols
  ASoC: sst-bxt-rt298: fix obsoleted initializers for array
  ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array
  ASoC: rt5616: add static qualifier for file local symbols
  ASoC: arizona: Add output power up/down delays for speaker path
  ASoC: arizona: Add debug prints for output power up/down times
  ALSA: hda - Add the top speaker pin config for HP Spectre x360
  ASoC: Intel: Add DMIC channel constraint for bxt machine
  ...
This commit is contained in:
Linus Torvalds 2016-10-05 12:13:14 -07:00
parent 878fb5dc96 eeea8b40cd
commit 5691f0e9a3
408 changed files with 16528 additions and 3996 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

16
Documentation/devicetree/bindings/sound/nau8810.txt Normal file
View File

@ -0,0 +1,16 @@
NAU8810 audio CODEC
This device supports I2C only.
Required properties:
- compatible : "nuvoton,nau8810"
- reg : the I2C address of the device.
Example:
codec: nau8810@1a {
compatible = "nuvoton,nau8810";
reg = <0x1a>;
};

42
Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt Normal file
View File

@ -0,0 +1,42 @@
NVIDIA Tegra audio complex, with SGTL5000 CODEC
Required properties:
- compatible : "nvidia,tegra-audio-sgtl5000"
- clocks : Must contain an entry for each entry in clock-names.
See ../clocks/clock-bindings.txt for details.
- clock-names : Must include the following entries:
- pll_a
- pll_a_out0
- mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
- 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 SGTL5000's pins (as documented in its binding), and the jacks
on the board:
* Headphone Jack
* Line In Jack
* Mic Jack
- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
connected to the CODEC.
- nvidia,audio-codec : The phandle of the SGTL5000 audio codec.
Example:
sound {
compatible = "toradex,tegra-audio-sgtl5000-apalis_t30",
"nvidia,tegra-audio-sgtl5000";
nvidia,model = "Toradex Apalis T30";
nvidia,audio-routing =
"Headphone Jack", "HP_OUT",
"LINE_IN", "Line In Jack",
"MIC_IN", "Mic Jack";
nvidia,i2s-controller = <&tegra_i2s2>;
nvidia,audio-codec = <&sgtl5000>;
clocks = <&tegra_car TEGRA30_CLK_PLL_A>,
<&tegra_car TEGRA30_CLK_PLL_A_OUT0>,
<&tegra_car TEGRA30_CLK_EXTERN1>;
clock-names = "pll_a", "pll_a_out0", "mclk";
};

30
Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt
View File

@ -16,6 +16,24 @@ Required properties:
* "spkr-iomux"
- qcom,model : Name of the sound card.
- qcom,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 could be power supplies, MicBias
of msm8x16_wcd codec and the jacks on the board:
Power supplies:
* MIC BIAS External1
* MIC BIAS External2
* MIC BIAS Internal1
* MIC BIAS Internal2
Board connectors:
* Headset Mic
* Secondary Mic",
* DMIC
* Ext Spk
Dai-link subnode properties and subnodes:
Required dai-link subnodes:
@ -37,6 +55,18 @@ sound: sound {
reg-names = "mic-iomux", "spkr-iomux";
qcom,model = "DB410c";
qcom,audio-routing =
"MIC BIAS External1", "Handset Mic",
"MIC BIAS Internal2", "Headset Mic",
"MIC BIAS External1", "Secondary Mic",
"AMIC1", "MIC BIAS External1",
"AMIC2", "MIC BIAS Internal2",
"AMIC3", "MIC BIAS External1",
"DMIC1", "MIC BIAS Internal1",
"MIC BIAS Internal1", "Digital Mic1",
"DMIC2", "MIC BIAS Internal1",
"MIC BIAS Internal1", "Digital Mic2";
/* I2S - Internal codec */
internal-dai-link@0 {
cpu { /* PRIMARY */

75
Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt
View File

@ -1,75 +0,0 @@
Renesas Sampling Rate Convert Sound Card:
Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC <-> codec.
Required properties:
- compatible : "renesas,rsrc-card{,<board>}"
Examples with boards are:
- "renesas,rsrc-card"
- "renesas,rsrc-card,lager"
- "renesas,rsrc-card,koelsch"
Optional properties:
- card_name : User specified audio sound card name, one string
property.
- cpu : CPU sub-node
- codec : CODEC sub-node
Optional subnode properties:
- format : CPU/CODEC common audio format.
"i2s", "right_j", "left_j" , "dsp_a"
"dsp_b", "ac97", "pdm", "msb", "lsb"
- frame-master : Indicates dai-link frame master.
phandle to a cpu or codec subnode.
- bitclock-master : Indicates dai-link bit clock master.
phandle to a cpu or codec subnode.
- bitclock-inversion : bool property. Add this if the
dai-link uses bit clock inversion.
- frame-inversion : bool property. Add this if the
dai-link uses frame clock inversion.
- convert-rate : platform specified sampling rate convert
- convert-channels : platform specified converted channel size (2 - 8 ch)
- audio-prefix : see audio-routing
- 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.
use audio-prefix if some components is using same sink/sources naming.
it can be used if compatible was "renesas,rsrc-card";
Required CPU/CODEC subnodes properties:
- sound-dai : phandle and port of CPU/CODEC
Optional CPU/CODEC subnodes properties:
- clocks / system-clock-frequency : specify subnode's clock if needed.
it can be specified via "clocks" if system has
clock node (= common clock), or "system-clock-frequency"
(if system doens't support common clock)
If a clock is specified, it is
enabled with clk_prepare_enable()
in dai startup() and disabled with
clk_disable_unprepare() in dai
shutdown().
Example
sound {
compatible = "renesas,rsrc-card,lager";
card-name = "rsnd-ak4643";
format = "left_j";
bitclock-master = <&sndcodec>;
frame-master = <&sndcodec>;
sndcpu: cpu {
sound-dai = <&rcar_sound>;
};
sndcodec: codec {
sound-dai = <&ak4643>;
system-clock-frequency = <11289600>;
};
};

22
Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt Normal file
View File

@ -0,0 +1,22 @@
ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards
Required properties:
- compatible: "rockchip,rk3399-gru-sound"
- rockchip,cpu: The phandle of the Rockchip I2S controller that's
connected to the codecs
- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs
Optional properties:
- dmic-wakeup-delay-ms : specify delay time (ms) for DMIC ready.
If this option is specified, which means it's required dmic need
delay for DMIC to ready so that rt5514 can avoid recording before
DMIC send valid data
Example:
sound {
compatible = "rockchip,rk3399-gru-sound";
rockchip,cpu = <&i2s0>;
rockchip,codec = <&max98357a &rt5514 &da7219>;
dmic-wakeup-delay-ms = <20>;
};

3
Documentation/devicetree/bindings/sound/rt5659.txt
View File

@ -12,6 +12,9 @@ Required properties:
Optional properties:
- clocks: The phandle of the master clock to the CODEC
- clock-names: Should be "mclk"
- realtek,in1-differential
- realtek,in3-differential
- realtek,in4-differential

47
Documentation/devicetree/bindings/sound/rt5660.txt Normal file
View File

@ -0,0 +1,47 @@
RT5660 audio CODEC
This device supports I2C only.
Required properties:
- compatible : "realtek,rt5660".
- reg : The I2C address of the device.
Optional properties:
- clocks: The phandle of the master clock to the CODEC
- clock-names: Should be "mclk"
- realtek,in1-differential
- realtek,in3-differential
Boolean. Indicate MIC1/3 input are differential, rather than single-ended.
- realtek,poweroff-in-suspend
Boolean. If the codec will be powered off in suspend, the resume should be
added delay time for waiting codec power ready.
- realtek,dmic1-data-pin
0: dmic1 is not used
1: using GPIO2 pin as dmic1 data pin
2: using IN1P pin as dmic1 data pin
Pins on the device (for linking into audio routes) for RT5660:
* DMIC L1
* DMIC R1
* IN1P
* IN1N
* IN2P
* IN3P
* IN3N
* SPO
* LOUTL
* LOUTR
Example:
rt5660 {
compatible = "realtek,rt5660";
reg = <0x1c>;
};

30
Documentation/devicetree/bindings/sound/rt5663.txt Normal file
View File

@ -0,0 +1,30 @@
RT5663/RT5668 audio CODEC
This device supports I2C only.
Required properties:
- compatible : One of "realtek,rt5663" or "realtek,rt5668".
- reg : The I2C address of the device.
- interrupts : The CODEC's interrupt output.
Optional properties:
Pins on the device (for linking into audio routes) for RT5663/RT5668:
* IN1P
* IN1N
* IN2P
* IN2N
* HPOL
* HPOR
Example:
codec: rt5663@12 {
compatible = "realtek,rt5663";
reg = <0x12>;
interrupts = <7 IRQ_TYPE_EDGE_FALLING>;
};

37
Documentation/devicetree/bindings/sound/simple-card.txt
View File

@ -22,6 +22,8 @@ Optional properties:
headphones are attached.
- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when
a microphone is attached.
- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such
as amplifiers, to be added to the sound card.
Optional subnodes:
@ -162,3 +164,38 @@ sound {
};
};
};
Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec
through TPA6130A2 amplifier to headphones:
&i2c0 {
codec: tlv320dac3100@18 {
compatible = "ti,tlv320dac3100";
...
}
amp: tpa6130a2@60 {
compatible = "ti,tpa6130a2";
...
}
}
sound {
compatible = "simple-audio-card";
...
simple-audio-card,widgets =
"Headphone", "Headphone Jack";
simple-audio-card,routing =
"Headphone Jack", "HPLEFT",
"Headphone Jack", "HPRIGHT",
"LEFTIN", "HPL",
"RIGHTIN", "HPR";
simple-audio-card,aux-devs = <&amp>;
simple-audio-card,cpu {
sound-dai = <&ssi2>;
};
simple-audio-card,codec {
sound-dai = <&codec>;
clocks = ...
};
};

110
Documentation/devicetree/bindings/sound/simple-scu-card.txt Normal file
View File

@ -0,0 +1,110 @@
ASoC simple SCU Sound Card
Simple-Card specifies audio DAI connections of SoC <-> codec.
Required properties:
- compatible : "simple-scu-audio-card"
"renesas,rsrc-card"
Optional properties:
- simple-audio-card,name : User specified audio sound card name, one string
property.
- simple-audio-card,cpu : CPU sub-node
- simple-audio-card,codec : CODEC sub-node
Optional subnode properties:
- simple-audio-card,format : CPU/CODEC common audio format.
"i2s", "right_j", "left_j" , "dsp_a"
"dsp_b", "ac97", "pdm", "msb", "lsb"
- simple-audio-card,frame-master : Indicates dai-link frame master.
phandle to a cpu or codec subnode.
- simple-audio-card,bitclock-master : Indicates dai-link bit clock master.
phandle to a cpu or codec subnode.
- simple-audio-card,bitclock-inversion : bool property. Add this if the
dai-link uses bit clock inversion.
- simple-audio-card,frame-inversion : bool property. Add this if the
dai-link uses frame clock inversion.
- simple-audio-card,convert-rate : platform specified sampling rate convert
- simple-audio-card,convert-channels : platform specified converted channel size (2 - 8 ch)
- simple-audio-card,prefix : see audio-routing
- simple-audio-card,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.
use audio-prefix if some components is using same sink/sources naming.
it can be used if compatible was "renesas,rsrc-card";
Required CPU/CODEC subnodes properties:
- sound-dai : phandle and port of CPU/CODEC
Optional CPU/CODEC subnodes properties:
- clocks / system-clock-frequency : specify subnode's clock if needed.
it can be specified via "clocks" if system has
clock node (= common clock), or "system-clock-frequency"
(if system doens't support common clock)
If a clock is specified, it is
enabled with clk_prepare_enable()
in dai startup() and disabled with
clk_disable_unprepare() in dai
shutdown().
Example 1. Sampling Rate Covert
sound {
compatible = "simple-scu-audio-card";
simple-audio-card,name = "rsnd-ak4643";
simple-audio-card,format = "left_j";
simple-audio-card,format = "left_j";
simple-audio-card,bitclock-master = <&sndcodec>;
simple-audio-card,frame-master = <&sndcodec>;
simple-audio-card,convert-rate = <48000>; /* see audio_clk_a */
simple-audio-card,prefix = "ak4642";
simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
"DAI0 Capture", "ak4642 Capture";
sndcpu: simple-audio-card,cpu {
sound-dai = <&rcar_sound>;
};
sndcodec: simple-audio-card,codec {
sound-dai = <&ak4643>;
system-clock-frequency = <11289600>;
};
};
Example 2. 2 CPU 1 Codec
sound {
compatible = "renesas,rsrc-card";
card-name = "rsnd-ak4643";
format = "left_j";
bitclock-master = <&dpcmcpu>;
frame-master = <&dpcmcpu>;
convert-rate = <48000>; /* see audio_clk_a */
audio-prefix = "ak4642";
audio-routing = "ak4642 Playback", "DAI0 Playback",
"ak4642 Playback", "DAI1 Playback";
dpcmcpu: cpu@0 {
sound-dai = <&rcar_sound 0>;
};
cpu@1 {
sound-dai = <&rcar_sound 1>;
};
codec {
sound-dai = <&ak4643>;
clocks = <&audio_clock>;
};
};

52
Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
View File

@ -11,7 +11,9 @@ Documentation/devicetree/bindings/sound/simple-card.txt.
---------------------------------------
Required properties:
- compatible: "st,sti-uni-player" or "st,sti-uni-reader"
- compatible: "st,stih407-uni-player-hdmi", "st,stih407-uni-player-pcm-out",
"st,stih407-uni-player-dac", "st,stih407-uni-player-spdif",
"st,stih407-uni-reader-pcm_in", "st,stih407-uni-reader-hdmi",
- st,syscfg: phandle to boot-device system configuration registers
@ -33,32 +35,24 @@ Required properties:
"tx" for "st,sti-uni-player" compatibility
"rx" for "st,sti-uni-reader" compatibility
- st,version: IP version integrated in SOC.
- dai-name: DAI name that describes the IP.
- st,mode: IP working mode depending on associated codec.
"HDMI" connected to HDMI codec and support IEC HDMI formats (player only).
"SPDIF" connected to SPDIF codec and support SPDIF formats (player only).
"PCM" PCM standard mode for I2S or TDM bus.
"TDM" TDM mode for TDM bus.
Required properties ("st,sti-uni-player" compatibility only):
- clocks: CPU_DAI IP clock source, listed in the same order than the
CPU_DAI properties.
- st,uniperiph-id: internal SOC IP instance ID.
Optional properties:
- pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for
external codecs connection.
- pinctrl-names: should contain only one value - "default".
- st,tdm-mode: to declare to set TDM mode for unireader and uniplayer IPs.
Only compartible with IPs in charge of the external I2S/TDM bus.
Should be declared depending on associated codec.
Example:
sti_uni_player1: sti-uni-player@1 {
compatible = "st,sti-uni-player";
sti_uni_player1: sti-uni-player@0x8D81000 {
compatible = "st,stih407-uni-player-hdmi";
status = "okay";
#sound-dai-cells = <0>;
st,syscfg = <&syscfg_core>;
@ -66,15 +60,12 @@ Example:
reg = <0x8D81000 0x158>;
interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>;
dmas = <&fdma0 3 0 1>;
st,dai-name = "Uni Player #1 (I2S)";
dma-names = "tx";
st,uniperiph-id = <1>;
st,version = <5>;
st,mode = "TDM";
st,tdm-mode = <1>;
};
sti_uni_player2: sti-uni-player@2 {
compatible = "st,sti-uni-player";
sti_uni_player2: sti-uni-player@0x8D82000 {
compatible = "st,stih407-uni-player-pcm-out";
status = "okay";
#sound-dai-cells = <0>;
st,syscfg = <&syscfg_core>;
@ -82,15 +73,11 @@ Example:
reg = <0x8D82000 0x158>;
interrupts = <GIC_SPI 86 IRQ_TYPE_NONE>;
dmas = <&fdma0 4 0 1>;
dai-name = "Uni Player #2 (DAC)";
dma-names = "tx";
st,uniperiph-id = <2>;
st,version = <5>;
st,mode = "PCM";
};
sti_uni_player3: sti-uni-player@3 {
compatible = "st,sti-uni-player";
sti_uni_player3: sti-uni-player@0x8D85000 {
compatible = "st,stih407-uni-player-spdif";
status = "okay";
#sound-dai-cells = <0>;
st,syscfg = <&syscfg_core>;
@ -99,14 +86,10 @@ Example:
interrupts = <GIC_SPI 89 IRQ_TYPE_NONE>;
dmas = <&fdma0 7 0 1>;
dma-names = "tx";
dai-name = "Uni Player #3 (SPDIF)";
st,uniperiph-id = <3>;
st,version = <5>;
st,mode = "SPDIF";
};
sti_uni_reader1: sti-uni-reader@1 {
compatible = "st,sti-uni-reader";
sti_uni_reader1: sti-uni-reader@0x8D84000 {
compatible = "st,stih407-uni-reader-hdmi";
status = "disabled";
#sound-dai-cells = <0>;
st,syscfg = <&syscfg_core>;
@ -114,9 +97,6 @@ Example:
interrupts = <GIC_SPI 88 IRQ_TYPE_NONE>;
dmas = <&fdma0 6 0 1>;
dma-names = "rx";
dai-name = "Uni Reader #1 (HDMI RX)";
st,version = <3>;
st,mode = "PCM";
};
2) sti-sas-codec: internal audio codec IPs driver

3
Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt
View File

@ -9,6 +9,7 @@ Required properties:
- compatible : should be one of the following:
- "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC
- "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC
- reg : Offset and length of the register set for the device.
@ -25,6 +26,8 @@ Required properties:
"apb" clock for the spdif bus.
"spdif" clock for spdif controller.
- resets : reset specifier for the ahb reset (A31 and newer only)
Example:
spdif: spdif@01c21000 {

9
Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
View File

@ -11,6 +11,7 @@ Required properties:
"ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
"ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
"ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
"ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP)
- reg - <int> - I2C slave address
- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
@ -37,9 +38,11 @@ CODEC output pins:
* MICBIAS
CODEC input pins:
* MIC1LP
* MIC1RP
* MIC1LM
* MIC1LP, devices with ADC
* MIC1RP, devices with ADC
* MIC1LM, devices with ADC
* AIN1, devices without ADC
* AIN2, devices without ADC
The pins can be used in referring sound node's audio-routing property.

19
arch/arm/mach-s3c24xx/mach-mini2440.c
View File

@ -497,9 +497,28 @@ static struct i2c_board_info mini2440_i2c_devs[] __initdata = {
},
};
static struct uda134x_platform_data s3c24xx_uda134x = {
.l3 = {
.gpio_clk = S3C2410_GPB(4),
.gpio_data = S3C2410_GPB(3),
.gpio_mode = S3C2410_GPB(2),
.use_gpios = 1,
.data_hold = 1,
.data_setup = 1,
.clock_high = 1,
.mode_hold = 1,
.mode = 1,
.mode_setup = 1,
},
.model = UDA134X_UDA1341,
};
static struct platform_device uda1340_codec = {
.name = "uda134x-codec",
.id = -1,
.dev = {
.platform_data = &s3c24xx_uda134x,
},
};
static struct platform_device *mini2440_devices[] __initdata = {

2
include/sound/da7219.h
View File

@ -34,6 +34,8 @@ enum da7219_mic_amp_in_sel {
struct da7219_aad_pdata;
struct da7219_pdata {
bool wakeup_source;
/* Mic */
enum da7219_micbias_voltage micbias_lvl;
enum da7219_mic_amp_in_sel mic_amp_in_sel;

36
include/sound/hda_register.h
View File

@ -89,6 +89,19 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
#define AZX_REG_SD_BDLPL 0x18
#define AZX_REG_SD_BDLPU 0x1c
/* GTS registers */
#define AZX_REG_LLCH 0x14
#define AZX_REG_GTS_BASE 0x520
#define AZX_REG_GTSCC (AZX_REG_GTS_BASE + 0x00)
#define AZX_REG_WALFCC (AZX_REG_GTS_BASE + 0x04)
#define AZX_REG_TSCCL (AZX_REG_GTS_BASE + 0x08)
#define AZX_REG_TSCCU (AZX_REG_GTS_BASE + 0x0C)
#define AZX_REG_LLPFOC (AZX_REG_GTS_BASE + 0x14)
#define AZX_REG_LLPCL (AZX_REG_GTS_BASE + 0x18)
#define AZX_REG_LLPCU (AZX_REG_GTS_BASE + 0x1C)
/* Haswell/Broadwell display HD-A controller Extended Mode registers */
#define AZX_REG_HSW_EM4 0x100c
#define AZX_REG_HSW_EM5 0x1010
@ -242,6 +255,29 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
/* Interval used to calculate the iterating register offset */
#define AZX_DRSM_INTERVAL 0x08
/* Global time synchronization registers */
#define GTSCC_TSCCD_MASK 0x80000000
#define GTSCC_TSCCD_SHIFT BIT(31)
#define GTSCC_TSCCI_MASK 0x20
#define GTSCC_CDMAS_DMA_DIR_SHIFT 4
#define WALFCC_CIF_MASK 0x1FF
#define WALFCC_FN_SHIFT 9
#define HDA_CLK_CYCLES_PER_FRAME 512
/*
* An error occurs near frame "rollover". The clocks in frame value indicates
* whether this error may have occurred. Here we use the value of 10. Please
* see the errata for the right number [<10]
*/
#define HDA_MAX_CYCLE_VALUE 499
#define HDA_MAX_CYCLE_OFFSET 10
#define HDA_MAX_CYCLE_READ_RETRY 10
#define TSCCU_CCU_SHIFT 32
#define LLPC_CCU_SHIFT 32
/*
* helpers to read the stream position
*/

13
include/sound/hdaudio.h
View File

@ -245,6 +245,12 @@ struct hdac_rb {
/*
* HD-audio bus base driver
*
* @ppcap: pp capabilities pointer
* @spbcap: SPIB capabilities pointer
* @mlcap: MultiLink capabilities pointer
* @gtscap: gts capabilities pointer
* @drsmcap: dma resume capabilities pointer
*/
struct hdac_bus {
struct device *dev;
@ -256,6 +262,12 @@ struct hdac_bus {
void __iomem *remap_addr;
int irq;
void __iomem *ppcap;
void __iomem *spbcap;
void __iomem *mlcap;
void __iomem *gtscap;
void __iomem *drsmcap;
/* codec linked list */
struct list_head codec_list;
unsigned int num_codecs;
@ -335,6 +347,7 @@ static inline void snd_hdac_codec_link_down(struct hdac_device *codec)
int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val);
int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr,
unsigned int *res);
int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus);
int snd_hdac_link_power(struct hdac_device *codec, bool enable);
bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset);

12
include/sound/hdaudio_ext.h
View File

@ -8,11 +8,6 @@
*
* @bus: hdac bus
* @num_streams: streams supported
* @ppcap: pp capabilities pointer
* @spbcap: SPIB capabilities pointer
* @mlcap: MultiLink capabilities pointer
* @gtscap: gts capabilities pointer
* @drsmcap: dma resume capabilities pointer
* @hlink_list: link list of HDA links
* @lock: lock for link mgmt
* @cmd_dma_state: state of cmd DMAs: CORB and RIRB
@ -22,12 +17,6 @@ struct hdac_ext_bus {
int num_streams;
int idx;
void __iomem *ppcap;
void __iomem *spbcap;
void __iomem *mlcap;
void __iomem *gtscap;
void __iomem *drsmcap;
struct list_head hlink_list;
struct mutex lock;
@ -54,7 +43,6 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus);
#define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \
HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data)
int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *sbus);
void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable);
void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable);

15
include/sound/l3.h
View File

@ -2,9 +2,15 @@
#define _L3_H_ 1
struct l3_pins {
void (*setdat)(int);
void (*setclk)(int);
void (*setmode)(int);
void (*setdat)(struct l3_pins *, int);
void (*setclk)(struct l3_pins *, int);
void (*setmode)(struct l3_pins *, int);
int gpio_data;
int gpio_clk;
int gpio_mode;
int use_gpios;
int data_hold;
int data_setup;
int clock_high;
@ -13,6 +19,9 @@ struct l3_pins {
int mode_setup;
};
struct device;
int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
int l3_set_gpio_ops(struct device *dev, struct l3_pins *adap);
#endif

31
include/sound/rt5660.h Normal file
View File

@ -0,0 +1,31 @@
/*
* linux/sound/rt5660.h -- Platform data for RT5660
*
* Copyright 2016 Realtek Semiconductor Corp.
* Author: Oder Chiou <oder_chiou@realtek.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.
*/
#ifndef __LINUX_SND_RT5660_H
#define __LINUX_SND_RT5660_H
enum rt5660_dmic1_data_pin {
RT5660_DMIC1_NULL,
RT5660_DMIC1_DATA_GPIO2,
RT5660_DMIC1_DATA_IN1P,
};
struct rt5660_platform_data {
/* IN1 & IN3 can optionally be differential */
bool in1_diff;
bool in3_diff;
bool use_ldo2;
bool poweroff_codec_in_suspend;
enum rt5660_dmic1_data_pin dmic1_data_pin;
};
#endif

1
include/sound/s3c24xx_uda134x.h
View File

@ -7,7 +7,6 @@ struct s3c24xx_uda134x_platform_data {
int l3_clk;
int l3_mode;
int l3_data;
void (*power) (int);
int model;
};

35
include/sound/simple_card_utils.h
View File

@ -27,10 +27,45 @@ int asoc_simple_card_parse_daifmt(struct device *dev,
struct device_node *codec,
char *prefix,
unsigned int *retfmt);
__printf(3, 4)
int asoc_simple_card_set_dailink_name(struct device *dev,
struct snd_soc_dai_link *dai_link,
const char *fmt, ...);
int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
char *prefix);
#define asoc_simple_card_parse_clk_cpu(node, dai_link, simple_dai) \
asoc_simple_card_parse_clk(node, dai_link->cpu_of_node, simple_dai)
#define asoc_simple_card_parse_clk_codec(node, dai_link, simple_dai) \
asoc_simple_card_parse_clk(node, dai_link->codec_of_node, simple_dai)
int asoc_simple_card_parse_clk(struct device_node *node,
struct device_node *dai_of_node,
struct asoc_simple_dai *simple_dai);
#define asoc_simple_card_parse_cpu(node, dai_link, \
list_name, cells_name, is_single_link) \
asoc_simple_card_parse_dai(node, &dai_link->cpu_of_node, \
&dai_link->cpu_dai_name, list_name, cells_name, is_single_link)
#define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name) \
asoc_simple_card_parse_dai(node, &dai_link->codec_of_node, \
&dai_link->codec_dai_name, list_name, cells_name, NULL)
#define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \
asoc_simple_card_parse_dai(node, &dai_link->platform_of_node, \
NULL, list_name, cells_name, NULL)
int asoc_simple_card_parse_dai(struct device_node *node,
struct device_node **endpoint_np,
const char **dai_name,
const char *list_name,
const char *cells_name,
int *is_single_links);
int asoc_simple_card_init_dai(struct snd_soc_dai *dai,
struct asoc_simple_dai *simple_dai);
int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link);
void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link,
int is_single_links);
int asoc_simple_card_clean_reference(struct snd_soc_card *card);
#endif /* __SIMPLE_CARD_CORE_H */

19
include/sound/soc.h
View File

@ -898,14 +898,6 @@ struct snd_soc_codec_driver {
int (*resume)(struct snd_soc_codec *);
struct snd_soc_component_driver component_driver;
/* Default control and setup, added after probe() is run */
const struct snd_kcontrol_new *controls;
int num_controls;
const struct snd_soc_dapm_widget *dapm_widgets;
int num_dapm_widgets;
const struct snd_soc_dapm_route *dapm_routes;
int num_dapm_routes;
/* codec wide operations */
int (*set_sysclk)(struct snd_soc_codec *codec,
int clk_id, int source, unsigned int freq, int dir);
@ -1547,17 +1539,6 @@ static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platfo
return snd_soc_component_get_drvdata(&platform->component);
}
static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd,
void *data)
{
dev_set_drvdata(rtd->dev, data);
}
static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd)
{
return dev_get_drvdata(rtd->dev);
}
static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
{
INIT_LIST_HEAD(&card->codec_dev_list);

78
include/sound/tlv.h
View File

@ -22,67 +22,39 @@
*
*/
/*
* TLV structure is right behind the struct snd_ctl_tlv:
* unsigned int type - see SNDRV_CTL_TLVT_*
* unsigned int length
* .... data aligned to sizeof(unsigned int), use
* block_length = (length + (sizeof(unsigned int) - 1)) &
* ~(sizeof(unsigned int) - 1)) ....
*/
#include <uapi/sound/tlv.h>
#define TLV_ITEM(type, ...) \
(type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__
#define TLV_LENGTH(...) \
((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
/* For historical reasons, these macros are aliases to the ones in UAPI. */
#define TLV_ITEM SNDRV_CTL_TLVD_ITEM
#define TLV_LENGTH SNDRV_CTL_TLVD_LENGTH
#define TLV_CONTAINER_ITEM(...) \
TLV_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
#define DECLARE_TLV_CONTAINER(name, ...) \
unsigned int name[] = { TLV_CONTAINER_ITEM(__VA_ARGS__) }
#define TLV_CONTAINER_ITEM SNDRV_CTL_TLVD_CONTAINER_ITEM
#define DECLARE_TLV_CONTAINER SNDRV_CTL_TLVD_DECLARE_CONTAINER
#define TLV_DB_SCALE_MASK 0xffff
#define TLV_DB_SCALE_MUTE 0x10000
#define TLV_DB_SCALE_ITEM(min, step, mute) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_SCALE, \
(min), \
((step) & TLV_DB_SCALE_MASK) | \
((mute) ? TLV_DB_SCALE_MUTE : 0))
#define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
#define TLV_DB_SCALE_MASK SNDRV_CTL_TLVD_DB_SCALE_MASK
#define TLV_DB_SCALE_MUTE SNDRV_CTL_TLVD_DB_SCALE_MUTE
#define TLV_DB_SCALE_ITEM SNDRV_CTL_TLVD_DB_SCALE_ITEM
#define DECLARE_TLV_DB_SCALE SNDRV_CTL_TLVD_DECLARE_DB_SCALE
/* dB scale specified with min/max values instead of step */
#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) }
#define TLV_DB_MINMAX_ITEM SNDRV_CTL_TLVD_DB_MINMAX_ITEM
#define TLV_DB_MINMAX_MUTE_ITEM SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM
#define DECLARE_TLV_DB_MINMAX SNDRV_CTL_TLVD_DECLARE_DB_MINMAX
#define DECLARE_TLV_DB_MINMAX_MUTE SNDRV_CTL_TLVD_DECLARE_DB_MINMAX_MUTE
/* linear volume between min_dB and max_dB (.01dB unit) */
#define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
#define DECLARE_TLV_DB_LINEAR(name, min_dB, max_dB) \
unsigned int name[] = { TLV_DB_LINEAR_ITEM(min_dB, max_dB) }
#define TLV_DB_LINEAR_ITEM SNDRV_CTL_TLVD_DB_LINEAR_ITEM
#define DECLARE_TLV_DB_LINEAR SNDRV_CTL_TLVD_DECLARE_DB_LINEAR
/* dB range container:
* Items in dB range container must be ordered by their values and by their
* dB values. This implies that larger values must correspond with larger
* dB values (which is also required for all other mixer controls).
#define TLV_DB_RANGE_ITEM SNDRV_CTL_TLVD_DB_RANGE_ITEM
#define DECLARE_TLV_DB_RANGE SNDRV_CTL_TLVD_DECLARE_DB_RANGE
#define TLV_DB_GAIN_MUTE SNDRV_CTL_TLVD_DB_GAIN_MUTE
/*
* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR.
* This is an old fasion and obsoleted by commit bf1d1c9b6179("ALSA: tlv: add
* DECLARE_TLV_DB_RANGE()").
*/
/* Each item is: <min> <max> <TLV> */
#define TLV_DB_RANGE_ITEM(...) \
TLV_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
#define DECLARE_TLV_DB_RANGE(name, ...) \
unsigned int name[] = { TLV_DB_RANGE_ITEM(__VA_ARGS__) }
/* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */
#define TLV_DB_RANGE_HEAD(num) \
#define TLV_DB_RANGE_HEAD(num) \
SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int)
#define TLV_DB_GAIN_MUTE -9999999
#endif /* __SOUND_TLV_H */

1
include/uapi/sound/Kbuild
View File

@ -13,3 +13,4 @@ header-y += sb16_csp.h
header-y += sfnt_info.h
header-y += tlv.h
header-y += usb_stream.h
header-y += snd_sst_tokens.h

35
include/uapi/sound/asoc.h
View File

@ -83,7 +83,7 @@
#define SND_SOC_TPLG_NUM_TEXTS 16
/* ABI version */
#define SND_SOC_TPLG_ABI_VERSION 0x4
#define SND_SOC_TPLG_ABI_VERSION 0x5
/* Max size of TLV data */
#define SND_SOC_TPLG_TLV_SIZE 32
@ -105,7 +105,8 @@
#define SND_SOC_TPLG_TYPE_CODEC_LINK 9
#define SND_SOC_TPLG_TYPE_BACKEND_LINK 10
#define SND_SOC_TPLG_TYPE_PDATA 11
#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_PDATA
#define SND_SOC_TPLG_TYPE_BE_DAI 12
#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_BE_DAI
/* vendor block IDs - please add new vendor types to end */
#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000
@ -124,6 +125,11 @@
#define SND_SOC_TPLG_TUPLE_TYPE_WORD 4
#define SND_SOC_TPLG_TUPLE_TYPE_SHORT 5
/* BE DAI flags */
#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES (1 << 0)
#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1)
#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2)
/*
* Block Header.
* This header precedes all object and object arrays below.
@ -251,6 +257,7 @@ struct snd_soc_tplg_stream_caps {
__le32 period_size_max; /* max period size bytes */
__le32 buffer_size_min; /* min buffer size bytes */
__le32 buffer_size_max; /* max buffer size bytes */
__le32 sig_bits; /* number of bits of content */
} __attribute__((packed));
/*
@ -285,6 +292,8 @@ struct snd_soc_tplg_manifest {
__le32 graph_elems; /* number of graph elements */
__le32 pcm_elems; /* number of PCM elements */
__le32 dai_link_elems; /* number of DAI link elements */
__le32 be_dai_elems; /* number of BE DAI elements */
__le32 reserved[20]; /* reserved for new ABI element types */
struct snd_soc_tplg_private priv;
} __attribute__((packed));
@ -450,4 +459,26 @@ struct snd_soc_tplg_link_config {
struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
__le32 num_streams; /* number of streams */
} __attribute__((packed));
/*
* Describes SW/FW specific features of BE DAI.
*
* File block representation for BE DAI :-
* +-----------------------------------+-----+
* | struct snd_soc_tplg_hdr | 1 |
* +-----------------------------------+-----+
* | struct snd_soc_tplg_be_dai | N |
* +-----------------------------------+-----+
*/
struct snd_soc_tplg_be_dai {
__le32 size; /* in bytes of this structure */
char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */
__le32 dai_id; /* unique ID - used to match */
__le32 playback; /* supports playback mode */
__le32 capture; /* supports capture mode */
struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */
__le32 flag_mask; /* bitmask of flags to configure */
__le32 flags; /* SND_SOC_TPLG_DAI_FLGBIT_* */
struct snd_soc_tplg_private priv;
} __attribute__((packed));
#endif

3
include/uapi/sound/asound.h
View File

@ -106,9 +106,10 @@ enum {
SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */
SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */
SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */
SNDRV_HWDEP_IFACE_LINE6, /* Line6 USB processors */
/* Don't forget to change the following: */
SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM
SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_LINE6
};
struct snd_hwdep_info {

214
include/uapi/sound/snd_sst_tokens.h Normal file
View File

@ -0,0 +1,214 @@
/*
* snd_sst_tokens.h - Intel SST tokens definition
*
* Copyright (C) 2016 Intel Corp
* Author: Shreyas NC <shreyas.nc@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as 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.
*/
#ifndef __SND_SST_TOKENS_H__
#define __SND_SST_TOKENS_H__
/**
* %SKL_TKN_UUID: Module UUID
*
* %SKL_TKN_U8_BLOCK_TYPE: Type of the private data block.Can be:
* tuples, bytes, short and words
*
* %SKL_TKN_U8_IN_PIN_TYPE: Input pin type,
* homogenous=0, heterogenous=1
*
* %SKL_TKN_U8_OUT_PIN_TYPE: Output pin type,
* homogenous=0, heterogenous=1
* %SKL_TKN_U8_DYN_IN_PIN: Configure Input pin dynamically
* if true
*
* %SKL_TKN_U8_DYN_OUT_PIN: Configure Output pin dynamically
* if true
*
* %SKL_TKN_U8_IN_QUEUE_COUNT: Store the number of Input pins
*
* %SKL_TKN_U8_OUT_QUEUE_COUNT: Store the number of Output pins
*
* %SKL_TKN_U8_TIME_SLOT: TDM slot number
*
* %SKL_TKN_U8_CORE_ID: Stores module affinity value.Can take
* the values:
* SKL_AFFINITY_CORE_0 = 0,
* SKL_AFFINITY_CORE_1,
* SKL_AFFINITY_CORE_MAX
*
* %SKL_TKN_U8_MOD_TYPE: Module type value.
*
* %SKL_TKN_U8_CONN_TYPE: Module connection type can be a FE,
* BE or NONE as defined :
* SKL_PIPE_CONN_TYPE_NONE = 0,
* SKL_PIPE_CONN_TYPE_FE = 1 (HOST_DMA)
* SKL_PIPE_CONN_TYPE_BE = 2 (LINK_DMA)
*
* %SKL_TKN_U8_DEV_TYPE: Type of device to which the module is
* connected
* Can take the values:
* SKL_DEVICE_BT = 0x0,
* SKL_DEVICE_DMIC = 0x1,
* SKL_DEVICE_I2S = 0x2,
* SKL_DEVICE_SLIMBUS = 0x3,
* SKL_DEVICE_HDALINK = 0x4,
* SKL_DEVICE_HDAHOST = 0x5,
* SKL_DEVICE_NONE
*
* %SKL_TKN_U8_HW_CONN_TYPE: Connection type of the HW to which the
* module is connected
* SKL_CONN_NONE = 0,
* SKL_CONN_SOURCE = 1,
* SKL_CONN_SINK = 2
*
* %SKL_TKN_U16_PIN_INST_ID: Stores the pin instance id
*
* %SKL_TKN_U16_MOD_INST_ID: Stores the mdule instance id
*
* %SKL_TKN_U32_MAX_MCPS: Module max mcps value
*
* %SKL_TKN_U32_MEM_PAGES: Module resource pages
*
* %SKL_TKN_U32_OBS: Stores Output Buffer size
*
* %SKL_TKN_U32_IBS: Stores input buffer size
*
* %SKL_TKN_U32_VBUS_ID: Module VBUS_ID. PDM=0, SSP0=0,
* SSP1=1,SSP2=2,
* SSP3=3, SSP4=4,
* SSP5=5, SSP6=6,INVALID
*
* %SKL_TKN_U32_PARAMS_FIXUP: Module Params fixup mask
* %SKL_TKN_U32_CONVERTER: Module params converter mask
* %SKL_TKN_U32_PIPE_ID: Stores the pipe id
*
* %SKL_TKN_U32_PIPE_CONN_TYPE: Type of the token to which the pipe is
* connected to. It can be
* SKL_PIPE_CONN_TYPE_NONE = 0,
* SKL_PIPE_CONN_TYPE_FE = 1 (HOST_DMA),
* SKL_PIPE_CONN_TYPE_BE = 2 (LINK_DMA),
*
* %SKL_TKN_U32_PIPE_PRIORITY: Pipe priority value
* %SKL_TKN_U32_PIPE_MEM_PGS: Pipe resource pages
*
* %SKL_TKN_U32_DIR_PIN_COUNT: Value for the direction to set input/output
* formats and the pin count.
* The first 4 bits have the direction
* value and the next 4 have
* the pin count value.
* SKL_DIR_IN = 0, SKL_DIR_OUT = 1.
* The input and output formats
* share the same set of tokens
* with the distinction between input
* and output made by reading direction
* token.
*
* %SKL_TKN_U32_FMT_CH: Supported channel count
*
* %SKL_TKN_U32_FMT_FREQ: Supported frequency/sample rate
*
* %SKL_TKN_U32_FMT_BIT_DEPTH: Supported container size
*
* %SKL_TKN_U32_FMT_SAMPLE_SIZE:Number of samples in the container
*
* %SKL_TKN_U32_FMT_CH_CONFIG: Supported channel configurations for the
* input/output.
*
* %SKL_TKN_U32_FMT_INTERLEAVE: Interleaving style which can be per
* channel or per sample. The values can be :
* SKL_INTERLEAVING_PER_CHANNEL = 0,
* SKL_INTERLEAVING_PER_SAMPLE = 1,
*
* %SKL_TKN_U32_FMT_SAMPLE_TYPE:
* Specifies the sample type. Can take the
* values: SKL_SAMPLE_TYPE_INT_MSB = 0,
* SKL_SAMPLE_TYPE_INT_LSB = 1,
* SKL_SAMPLE_TYPE_INT_SIGNED = 2,
* SKL_SAMPLE_TYPE_INT_UNSIGNED = 3,
* SKL_SAMPLE_TYPE_FLOAT = 4
*
* %SKL_TKN_U32_CH_MAP: Channel map values
* %SKL_TKN_U32_MOD_SET_PARAMS: It can take these values:
* SKL_PARAM_DEFAULT, SKL_PARAM_INIT,
* SKL_PARAM_SET, SKL_PARAM_BIND
*
* %SKL_TKN_U32_MOD_PARAM_ID: ID of the module params
*
* %SKL_TKN_U32_CAPS_SET_PARAMS:
* Set params value
*
* %SKL_TKN_U32_CAPS_PARAMS_ID: Params ID
*
* %SKL_TKN_U32_CAPS_SIZE: Caps size
*
* %SKL_TKN_U32_PROC_DOMAIN: Specify processing domain
*
* %SKL_TKN_U32_LIB_COUNT: Specifies the number of libraries
*
* %SKL_TKN_STR_LIB_NAME: Specifies the library name
*
* module_id and loadable flags dont have tokens as these values will be
* read from the DSP FW manifest
*/
enum SKL_TKNS {
SKL_TKN_UUID = 1,
SKL_TKN_U8_NUM_BLOCKS,
SKL_TKN_U8_BLOCK_TYPE,
SKL_TKN_U8_IN_PIN_TYPE,
SKL_TKN_U8_OUT_PIN_TYPE,
SKL_TKN_U8_DYN_IN_PIN,
SKL_TKN_U8_DYN_OUT_PIN,
SKL_TKN_U8_IN_QUEUE_COUNT,
SKL_TKN_U8_OUT_QUEUE_COUNT,
SKL_TKN_U8_TIME_SLOT,
SKL_TKN_U8_CORE_ID,
SKL_TKN_U8_MOD_TYPE,
SKL_TKN_U8_CONN_TYPE,
SKL_TKN_U8_DEV_TYPE,
SKL_TKN_U8_HW_CONN_TYPE,
SKL_TKN_U16_MOD_INST_ID,
SKL_TKN_U16_BLOCK_SIZE,
SKL_TKN_U32_MAX_MCPS,
SKL_TKN_U32_MEM_PAGES,
SKL_TKN_U32_OBS,
SKL_TKN_U32_IBS,
SKL_TKN_U32_VBUS_ID,
SKL_TKN_U32_PARAMS_FIXUP,
SKL_TKN_U32_CONVERTER,
SKL_TKN_U32_PIPE_ID,
SKL_TKN_U32_PIPE_CONN_TYPE,
SKL_TKN_U32_PIPE_PRIORITY,
SKL_TKN_U32_PIPE_MEM_PGS,
SKL_TKN_U32_DIR_PIN_COUNT,
SKL_TKN_U32_FMT_CH,
SKL_TKN_U32_FMT_FREQ,
SKL_TKN_U32_FMT_BIT_DEPTH,
SKL_TKN_U32_FMT_SAMPLE_SIZE,
SKL_TKN_U32_FMT_CH_CONFIG,
SKL_TKN_U32_FMT_INTERLEAVE,
SKL_TKN_U32_FMT_SAMPLE_TYPE,
SKL_TKN_U32_FMT_CH_MAP,
SKL_TKN_U32_PIN_MOD_ID,
SKL_TKN_U32_PIN_INST_ID,
SKL_TKN_U32_MOD_SET_PARAMS,
SKL_TKN_U32_MOD_PARAM_ID,
SKL_TKN_U32_CAPS_SET_PARAMS,
SKL_TKN_U32_CAPS_PARAMS_ID,
SKL_TKN_U32_CAPS_SIZE,
SKL_TKN_U32_PROC_DOMAIN,
SKL_TKN_U32_LIB_COUNT,
SKL_TKN_STR_LIB_NAME,
SKL_TKN_MAX = SKL_TKN_STR_LIB_NAME,
};
#endif

69
include/uapi/sound/tlv.h
View File

@ -28,4 +28,73 @@
#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */
#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */
/*
* TLV structure is right behind the struct snd_ctl_tlv:
* unsigned int type - see SNDRV_CTL_TLVT_*
* unsigned int length
* .... data aligned to sizeof(unsigned int), use
* block_length = (length + (sizeof(unsigned int) - 1)) &
* ~(sizeof(unsigned int) - 1)) ....
*/
#define SNDRV_CTL_TLVD_ITEM(type, ...) \
(type), SNDRV_CTL_TLVD_LENGTH(__VA_ARGS__), __VA_ARGS__
#define SNDRV_CTL_TLVD_LENGTH(...) \
((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
#define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
#define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \
unsigned int name[] = { \
SNDRV_CTL_TLVD_CONTAINER_ITEM(__VA_ARGS__) \
}
#define SNDRV_CTL_TLVD_DB_SCALE_MASK 0xffff
#define SNDRV_CTL_TLVD_DB_SCALE_MUTE 0x10000
#define SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_SCALE, \
(min), \
((step) & SNDRV_CTL_TLVD_DB_SCALE_MASK) | \
((mute) ? SNDRV_CTL_TLVD_DB_SCALE_MUTE : 0))
#define SNDRV_CTL_TLVD_DECLARE_DB_SCALE(name, min, step, mute) \
unsigned int name[] = { \
SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
}
/* dB scale specified with min/max values instead of step */
#define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
#define SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
#define SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(name, min_dB, max_dB) \
unsigned int name[] = { \
SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
}
#define SNDRV_CTL_TLVD_DECLARE_DB_MINMAX_MUTE(name, min_dB, max_dB) \
unsigned int name[] = { \
SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
}
/* linear volume between min_dB and max_dB (.01dB unit) */
#define SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
#define SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(name, min_dB, max_dB) \
unsigned int name[] = { \
SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
}
/* dB range container:
* Items in dB range container must be ordered by their values and by their
* dB values. This implies that larger values must correspond with larger
* dB values (which is also required for all other mixer controls).
*/
/* Each item is: <min> <max> <TLV> */
#define SNDRV_CTL_TLVD_DB_RANGE_ITEM(...) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
#define SNDRV_CTL_TLVD_DECLARE_DB_RANGE(name, ...) \
unsigned int name[] = { \
SNDRV_CTL_TLVD_DB_RANGE_ITEM(__VA_ARGS__) \
}
#define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999
#endif

15
sound/aoa/fabrics/layout.c
View File

@ -112,6 +112,7 @@ MODULE_ALIAS("sound-layout-100");
MODULE_ALIAS("aoa-device-id-14");
MODULE_ALIAS("aoa-device-id-22");
MODULE_ALIAS("aoa-device-id-31");
MODULE_ALIAS("aoa-device-id-35");
MODULE_ALIAS("aoa-device-id-44");
@ -362,6 +363,13 @@ static struct layout layouts[] = {
.connections = tas_connections_nolineout,
},
},
/* PowerBook6,1 */
{ .device_id = 31,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_nolineout,
},
},
/* PowerBook6,5 */
{ .device_id = 44,
.codecs[0] = {
@ -1161,12 +1169,7 @@ static struct soundbus_driver aoa_soundbus_driver = {
static int __init aoa_fabric_layout_init(void)
{
int err;
err = soundbus_register_driver(&aoa_soundbus_driver);
if (err)
return err;
return 0;
return soundbus_register_driver(&aoa_soundbus_driver);
}
static void __exit aoa_fabric_layout_exit(void)

2
sound/aoa/soundbus/i2sbus/core.c
View File

@ -197,7 +197,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
* so restrict to those we do handle for now.
*/
if (id && (*id == 22 || *id == 14 || *id == 35 ||
*id == 44)) {
*id == 31 || *id == 44)) {
snprintf(dev->sound.modalias, 32,
"aoa-device-id-%d", *id);
ok = 1;

14
sound/core/compress_offload.c
View File

@ -553,13 +553,9 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
* we should allow parameter change only when stream has been
* opened not in other cases
*/
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;
if (copy_from_user(params, (void __user *)arg, sizeof(*params))) {
retval = -EFAULT;
goto out;
}
params = memdup_user((void __user *)arg, sizeof(*params));
if (IS_ERR(params))
return PTR_ERR(params);
retval = snd_compress_check_input(params);
if (retval)
@ -784,7 +780,7 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream)
ret = wait_event_interruptible(stream->runtime->sleep,
(stream->runtime->state != SNDRV_PCM_STATE_DRAINING));
if (ret == -ERESTARTSYS)
pr_debug("wait aborted by a signal");
pr_debug("wait aborted by a signal\n");
else if (ret)
pr_debug("wait for drain failed with %d\n", ret);
@ -966,7 +962,7 @@ static int snd_compress_dev_register(struct snd_device *device)
compr->card, compr->device,
&snd_compr_file_ops, compr, &compr->dev);
if (ret < 0) {
pr_err("snd_register_device failed\n %d", ret);
pr_err("snd_register_device failed %d\n", ret);
return ret;
}
return ret;

4
sound/core/pcm_native.c
View File

@ -2637,9 +2637,11 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream)
break;
/* Fall through */
case SNDRV_PCM_STATE_PREPARED:
case SNDRV_PCM_STATE_SUSPENDED:
err = 0;
break;
case SNDRV_PCM_STATE_SUSPENDED:
err = -ESTRPIPE;
break;
case SNDRV_PCM_STATE_XRUN:
err = -EPIPE;
break;

724
sound/core/seq/seq_clientmgr.c
View File

File diff suppressed because it is too large Load Diff

7
sound/core/seq/seq_compat.c
View File

@ -47,7 +47,6 @@ static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned
{
int err = -EFAULT;
struct snd_seq_port_info *data;
mm_segment_t fs;
data = kmalloc(sizeof(*data), GFP_KERNEL);
if (!data)
@ -59,9 +58,7 @@ static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned
goto error;
data->kernel = NULL;
fs = snd_enter_user();
err = snd_seq_do_ioctl(client, cmd, data);
snd_leave_user(fs);
err = snd_seq_kernel_client_ctl(client->number, cmd, &data);
if (err < 0)
goto error;
@ -123,7 +120,7 @@ static long snd_seq_ioctl_compat(struct file *file, unsigned int cmd, unsigned l
case SNDRV_SEQ_IOCTL_GET_SUBSCRIPTION:
case SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT:
case SNDRV_SEQ_IOCTL_RUNNING_MODE:
return snd_seq_do_ioctl(client, cmd, argp);
return snd_seq_ioctl(file, cmd, arg);
case SNDRV_SEQ_IOCTL_CREATE_PORT32:
return snd_seq_call_port_info_ioctl(client, SNDRV_SEQ_IOCTL_CREATE_PORT, argp);
case SNDRV_SEQ_IOCTL_DELETE_PORT32:

5
sound/firewire/bebob/Makefile
View File

@ -1,4 +1,5 @@
snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \
bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \
bebob_focusrite.o bebob_maudio.o bebob.o
bebob_pcm.o bebob_hwdep.o bebob_terratec.o \
bebob_yamaha_terratec.o bebob_focusrite.o bebob_maudio.o \
bebob.o
obj-$(CONFIG_SND_BEBOB) += snd-bebob.o

8
sound/firewire/bebob/bebob.c
View File

@ -458,17 +458,17 @@ static const struct ieee1394_device_id bebob_id_table[] = {
/* TerraTec Electronic GmbH, PHASE 88 Rack FW */
SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000003, &phase88_rack_spec),
/* TerraTec Electronic GmbH, PHASE 24 FW */
SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000004, &phase24_series_spec),
SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000004, &yamaha_terratec_spec),
/* TerraTec Electronic GmbH, Phase X24 FW */
SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000007, &phase24_series_spec),
SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000007, &yamaha_terratec_spec),
/* TerraTec Electronic GmbH, EWS MIC2/MIC8 */
SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000005, &spec_normal),
/* Terratec Electronic GmbH, Aureon 7.1 Firewire */
SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000002, &spec_normal),
/* Yamaha, GO44 */
SND_BEBOB_DEV_ENTRY(VEN_YAMAHA, 0x0010000b, &yamaha_go_spec),
SND_BEBOB_DEV_ENTRY(VEN_YAMAHA, 0x0010000b, &yamaha_terratec_spec),
/* YAMAHA, GO46 */
SND_BEBOB_DEV_ENTRY(VEN_YAMAHA, 0x0010000c, &yamaha_go_spec),
SND_BEBOB_DEV_ENTRY(VEN_YAMAHA, 0x0010000c, &yamaha_terratec_spec),
/* Focusrite, SaffirePro 26 I/O */
SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec),
/* Focusrite, SaffirePro 10 I/O */

3
sound/firewire/bebob/bebob.h
View File

@ -235,8 +235,7 @@ int snd_bebob_create_hwdep_device(struct snd_bebob *bebob);
/* model specific operations */
extern const struct snd_bebob_spec phase88_rack_spec;
extern const struct snd_bebob_spec phase24_series_spec;
extern const struct snd_bebob_spec yamaha_go_spec;
extern const struct snd_bebob_spec yamaha_terratec_spec;
extern const struct snd_bebob_spec saffirepro_26_spec;
extern const struct snd_bebob_spec saffirepro_10_spec;
extern const struct snd_bebob_spec saffire_le_spec;

31
sound/firewire/bebob/bebob_terratec.c
View File

@ -36,25 +36,6 @@ end:
return err;
}
static enum snd_bebob_clock_type phase24_series_clk_src_types[] = {
SND_BEBOB_CLOCK_TYPE_INTERNAL,
SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */
};
static int
phase24_series_clk_src_get(struct snd_bebob *bebob, unsigned int *id)
{
int err;
err = avc_audio_get_selector(bebob->unit, 0, 4, id);
if (err < 0)
return err;
if (*id >= ARRAY_SIZE(phase24_series_clk_src_types))
return -EIO;
return 0;
}
static const struct snd_bebob_rate_spec phase_series_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
@ -71,15 +52,3 @@ const struct snd_bebob_spec phase88_rack_spec = {
.rate = &phase_series_rate_spec,
.meter = NULL
};
/* 'PHASE 24 FW' and 'PHASE X24 FW' */
static const struct snd_bebob_clock_spec phase24_series_clk = {
.num = ARRAY_SIZE(phase24_series_clk_src_types),
.types = phase24_series_clk_src_types,
.get = &phase24_series_clk_src_get,
};
const struct snd_bebob_spec phase24_series_spec = {
.clock = &phase24_series_clk,
.rate = &phase_series_rate_spec,
.meter = NULL
};

9
sound/firewire/bebob/bebob_yamaha.c → sound/firewire/bebob/bebob_yamaha_terratec.c
View File

@ -14,7 +14,7 @@
* must be accompanied. If changing the state, a LED on the device starts to
* blink and its sync status is false. In this state, the device sounds nothing
* even if streaming. To start streaming at the current sampling rate is only
* way to revocer this state. GO46 is better for stand-alone mixer.
* way to recover this state. GO46 is better for stand-alone mixer.
*
* Both of them have a capability to change its sampling rate up to 192.0kHz.
* At 192.0kHz, the device reports 4 PCM-in, 1 MIDI-in, 6 PCM-out, 1 MIDI-out.
@ -25,7 +25,10 @@
* streaming with many asynchronous transactions brings sounds with noises.
* Unfortunately current 'ffado-mixer' generated many asynchronous transaction
* to observe device's state, mainly check cmp connection and signal format. I
* reccomend users to close ffado-mixer at 192.0kHz if mixer is needless.
* recommend users to close ffado-mixer at 192.0kHz if mixer is needless.
*
* Terratec PHASE 24 FW and PHASE X24 FW are internally the same as
* Yamaha GO 44 and GO 46. Yamaha and Terratec had cooperated for these models.
*/
static enum snd_bebob_clock_type clk_src_types[] = {
@ -55,7 +58,7 @@ static const struct snd_bebob_rate_spec rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
const struct snd_bebob_spec yamaha_go_spec = {
const struct snd_bebob_spec yamaha_terratec_spec = {
.clock = &clock_spec,
.rate = &rate_spec,
.meter = NULL

4
sound/firewire/dice/dice-pcm.c
View File

@ -302,7 +302,7 @@ static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream)
int snd_dice_create_pcm(struct snd_dice *dice)
{
static struct snd_pcm_ops capture_ops = {
static const struct snd_pcm_ops capture_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -314,7 +314,7 @@ int snd_dice_create_pcm(struct snd_dice *dice)
.page = snd_pcm_lib_get_vmalloc_page,
.mmap = snd_pcm_lib_mmap_vmalloc,
};
static struct snd_pcm_ops playback_ops = {
static const struct snd_pcm_ops playback_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/firewire/digi00x/digi00x-pcm.c
View File

@ -329,7 +329,7 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
return amdtp_stream_pcm_pointer(&dg00x->rx_stream);
}
static struct snd_pcm_ops pcm_capture_ops = {
static const struct snd_pcm_ops pcm_capture_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -341,7 +341,7 @@ static struct snd_pcm_ops pcm_capture_ops = {
.page = snd_pcm_lib_get_vmalloc_page,
};
static struct snd_pcm_ops pcm_playback_ops = {
static const struct snd_pcm_ops pcm_playback_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/firewire/oxfw/oxfw-pcm.c
View File

@ -388,7 +388,7 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstm)
int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
{
static struct snd_pcm_ops capture_ops = {
static const struct snd_pcm_ops capture_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -400,7 +400,7 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
.page = snd_pcm_lib_get_vmalloc_page,
.mmap = snd_pcm_lib_mmap_vmalloc,
};
static struct snd_pcm_ops playback_ops = {
static const struct snd_pcm_ops playback_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/firewire/tascam/tascam-pcm.c
View File

@ -268,7 +268,7 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
return amdtp_stream_pcm_pointer(&tscm->rx_stream);
}
static struct snd_pcm_ops pcm_capture_ops = {
static const struct snd_pcm_ops pcm_capture_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -280,7 +280,7 @@ static struct snd_pcm_ops pcm_capture_ops = {
.page = snd_pcm_lib_get_vmalloc_page,
};
static struct snd_pcm_ops pcm_playback_ops = {
static const struct snd_pcm_ops pcm_playback_ops = {
.open = pcm_open,
.close = pcm_close,
.ioctl = snd_pcm_lib_ioctl,

91
sound/hda/ext/hdac_ext_controller.c
View File

@ -29,81 +29,6 @@
*/
#define HDAC_MAX_CAPS 10
/**
* snd_hdac_ext_bus_parse_capabilities - parse capablity structure
* @ebus: the pointer to extended bus object
*
* Returns 0 if successful, or a negative error code.
*/
int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *ebus)
{
unsigned int cur_cap;
unsigned int offset;
struct hdac_bus *bus = &ebus->bus;
unsigned int counter = 0;
offset = snd_hdac_chip_readl(bus, LLCH);
/* Lets walk the linked capabilities list */
do {
cur_cap = _snd_hdac_chip_read(l, bus, offset);
dev_dbg(bus->dev, "Capability version: 0x%x\n",
((cur_cap & AZX_CAP_HDR_VER_MASK) >> AZX_CAP_HDR_VER_OFF));
dev_dbg(bus->dev, "HDA capability ID: 0x%x\n",
(cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF);
switch ((cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF) {
case AZX_ML_CAP_ID:
dev_dbg(bus->dev, "Found ML capability\n");
ebus->mlcap = bus->remap_addr + offset;
break;
case AZX_GTS_CAP_ID:
dev_dbg(bus->dev, "Found GTS capability offset=%x\n", offset);
ebus->gtscap = bus->remap_addr + offset;
break;
case AZX_PP_CAP_ID:
/* PP capability found, the Audio DSP is present */
dev_dbg(bus->dev, "Found PP capability offset=%x\n", offset);
ebus->ppcap = bus->remap_addr + offset;
break;
case AZX_SPB_CAP_ID:
/* SPIB capability found, handler function */
dev_dbg(bus->dev, "Found SPB capability\n");
ebus->spbcap = bus->remap_addr + offset;
break;
case AZX_DRSM_CAP_ID:
/* DMA resume capability found, handler function */
dev_dbg(bus->dev, "Found DRSM capability\n");
ebus->drsmcap = bus->remap_addr + offset;
break;
default:
dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap);
break;
}
counter++;
if (counter > HDAC_MAX_CAPS) {
dev_err(bus->dev, "We exceeded HDAC Ext capablities!!!\n");
break;
}
/* read the offset of next capabiity */
offset = cur_cap & AZX_CAP_HDR_NXT_PTR_MASK;
} while (offset);
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_parse_capabilities);
/*
* processing pipe helpers - these helpers are useful for dealing with HDA
* new capability of processing pipelines
@ -118,15 +43,15 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *ebus, bool enable)
{
struct hdac_bus *bus = &ebus->bus;
if (!ebus->ppcap) {
if (!bus->ppcap) {
dev_err(bus->dev, "Address of PP capability is NULL");
return;
}
if (enable)
snd_hdac_updatel(ebus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN);
snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN);
else
snd_hdac_updatel(ebus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0);
snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable);
@ -139,15 +64,15 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *ebus, bool enable)
{
struct hdac_bus *bus = &ebus->bus;
if (!ebus->ppcap) {
if (!bus->ppcap) {
dev_err(bus->dev, "Address of PP capability is NULL\n");
return;
}
if (enable)
snd_hdac_updatel(ebus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE);
snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE);
else
snd_hdac_updatel(ebus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0);
snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable);
@ -171,7 +96,7 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
struct hdac_ext_link *hlink;
struct hdac_bus *bus = &ebus->bus;
link_count = readl(ebus->mlcap + AZX_REG_ML_MLCD) + 1;
link_count = readl(bus->mlcap + AZX_REG_ML_MLCD) + 1;
dev_dbg(bus->dev, "In %s Link count: %d\n", __func__, link_count);
@ -181,7 +106,7 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
return -ENOMEM;
hlink->index = idx;
hlink->bus = bus;
hlink->ml_addr = ebus->mlcap + AZX_ML_BASE +
hlink->ml_addr = bus->mlcap + AZX_ML_BASE +
(AZX_ML_INTERVAL * idx);
hlink->lcaps = readl(hlink->ml_addr + AZX_REG_ML_LCAP);
hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID);

56
sound/hda/ext/hdac_ext_stream.c
View File

@ -40,27 +40,27 @@ void snd_hdac_ext_stream_init(struct hdac_ext_bus *ebus,
{
struct hdac_bus *bus = &ebus->bus;
if (ebus->ppcap) {
stream->pphc_addr = ebus->ppcap + AZX_PPHC_BASE +
if (bus->ppcap) {
stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE +
AZX_PPHC_INTERVAL * idx;
stream->pplc_addr = ebus->ppcap + AZX_PPLC_BASE +
stream->pplc_addr = bus->ppcap + AZX_PPLC_BASE +
AZX_PPLC_MULTI * ebus->num_streams +
AZX_PPLC_INTERVAL * idx;
}
if (ebus->spbcap) {
stream->spib_addr = ebus->spbcap + AZX_SPB_BASE +
if (bus->spbcap) {
stream->spib_addr = bus->spbcap + AZX_SPB_BASE +
AZX_SPB_INTERVAL * idx +
AZX_SPB_SPIB;
stream->fifo_addr = ebus->spbcap + AZX_SPB_BASE +
stream->fifo_addr = bus->spbcap + AZX_SPB_BASE +
AZX_SPB_INTERVAL * idx +
AZX_SPB_MAXFIFO;
}
if (ebus->drsmcap)
stream->dpibr_addr = ebus->drsmcap + AZX_DRSM_BASE +
if (bus->drsmcap)
stream->dpibr_addr = bus->drsmcap + AZX_DRSM_BASE +
AZX_DRSM_INTERVAL * idx;
stream->decoupled = false;
@ -131,10 +131,10 @@ void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *ebus,
spin_lock_irq(&bus->reg_lock);
if (decouple)
snd_hdac_updatel(ebus->ppcap, AZX_REG_PP_PPCTL, 0,
snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0,
AZX_PPCTL_PROCEN(hstream->index));
else
snd_hdac_updatel(ebus->ppcap, AZX_REG_PP_PPCTL,
snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
AZX_PPCTL_PROCEN(hstream->index), 0);
stream->decoupled = decouple;
spin_unlock_irq(&bus->reg_lock);
@ -255,7 +255,7 @@ hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus,
struct hdac_stream *stream = NULL;
struct hdac_bus *hbus = &ebus->bus;
if (!ebus->ppcap) {
if (!hbus->ppcap) {
dev_err(hbus->dev, "stream type not supported\n");
return NULL;
}
@ -296,7 +296,7 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus,
struct hdac_stream *stream = NULL;
struct hdac_bus *hbus = &ebus->bus;
if (!ebus->ppcap) {
if (!hbus->ppcap) {
dev_err(hbus->dev, "stream type not supported\n");
return NULL;
}
@ -423,21 +423,21 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *ebus,
u32 register_mask = 0;
struct hdac_bus *bus = &ebus->bus;
if (!ebus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL");
if (!bus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL\n");
return;
}
mask |= (1 << index);
register_mask = readl(ebus->spbcap + AZX_REG_SPB_SPBFCCTL);
register_mask = readl(bus->spbcap + AZX_REG_SPB_SPBFCCTL);
mask |= register_mask;
if (enable)
snd_hdac_updatel(ebus->spbcap, AZX_REG_SPB_SPBFCCTL, 0, mask);
snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, 0, mask);
else
snd_hdac_updatel(ebus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, 0);
snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_spbcap_enable);
@ -452,8 +452,8 @@ int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus,
{
struct hdac_bus *bus = &ebus->bus;
if (!ebus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL");
if (!bus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL\n");
return -EINVAL;
}
@ -475,8 +475,8 @@ int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus,
{
struct hdac_bus *bus = &ebus->bus;
if (!ebus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL");
if (!bus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL\n");
return -EINVAL;
}
@ -515,21 +515,21 @@ void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus,
u32 register_mask = 0;
struct hdac_bus *bus = &ebus->bus;
if (!ebus->drsmcap) {
dev_err(bus->dev, "Address of DRSM capability is NULL");
if (!bus->drsmcap) {
dev_err(bus->dev, "Address of DRSM capability is NULL\n");
return;
}
mask |= (1 << index);
register_mask = readl(ebus->drsmcap + AZX_REG_SPB_SPBFCCTL);
register_mask = readl(bus->drsmcap + AZX_REG_SPB_SPBFCCTL);
mask |= register_mask;
if (enable)
snd_hdac_updatel(ebus->drsmcap, AZX_REG_DRSM_CTL, 0, mask);
snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, 0, mask);
else
snd_hdac_updatel(ebus->drsmcap, AZX_REG_DRSM_CTL, mask, 0);
snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_drsm_enable);
@ -544,8 +544,8 @@ int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus,
{
struct hdac_bus *bus = &ebus->bus;
if (!ebus->drsmcap) {
dev_err(bus->dev, "Address of DRSM capability is NULL");
if (!bus->drsmcap) {
dev_err(bus->dev, "Address of DRSM capability is NULL\n");
return -EINVAL;
}

75
sound/hda/hdac_controller.c
View File

@ -255,6 +255,81 @@ int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr,
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_get_response);
#define HDAC_MAX_CAPS 10
/**
* snd_hdac_bus_parse_capabilities - parse capability structure
* @bus: the pointer to bus object
*
* Returns 0 if successful, or a negative error code.
*/
int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus)
{
unsigned int cur_cap;
unsigned int offset;
unsigned int counter = 0;
offset = snd_hdac_chip_readl(bus, LLCH);
/* Lets walk the linked capabilities list */
do {
cur_cap = _snd_hdac_chip_read(l, bus, offset);
dev_dbg(bus->dev, "Capability version: 0x%x\n",
(cur_cap & AZX_CAP_HDR_VER_MASK) >> AZX_CAP_HDR_VER_OFF);
dev_dbg(bus->dev, "HDA capability ID: 0x%x\n",
(cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF);
switch ((cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF) {
case AZX_ML_CAP_ID:
dev_dbg(bus->dev, "Found ML capability\n");
bus->mlcap = bus->remap_addr + offset;
break;
case AZX_GTS_CAP_ID:
dev_dbg(bus->dev, "Found GTS capability offset=%x\n", offset);
bus->gtscap = bus->remap_addr + offset;
break;
case AZX_PP_CAP_ID:
/* PP capability found, the Audio DSP is present */
dev_dbg(bus->dev, "Found PP capability offset=%x\n", offset);
bus->ppcap = bus->remap_addr + offset;
break;
case AZX_SPB_CAP_ID:
/* SPIB capability found, handler function */
dev_dbg(bus->dev, "Found SPB capability\n");
bus->spbcap = bus->remap_addr + offset;
break;
case AZX_DRSM_CAP_ID:
/* DMA resume capability found, handler function */
dev_dbg(bus->dev, "Found DRSM capability\n");
bus->drsmcap = bus->remap_addr + offset;
break;
default:
dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap);
break;
}
counter++;
if (counter > HDAC_MAX_CAPS) {
dev_err(bus->dev, "We exceeded HDAC capabilities!!!\n");
break;
}
/* read the offset of next capability */
offset = cur_cap & AZX_CAP_HDR_NXT_PTR_MASK;
} while (offset);
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_parse_capabilities);
/*
* Lowlevel interface
*/

4
sound/pci/ad1889.c
View File

@ -571,7 +571,7 @@ snd_ad1889_capture_pointer(struct snd_pcm_substream *ss)
return bytes_to_frames(ss->runtime, ptr);
}
static struct snd_pcm_ops snd_ad1889_playback_ops = {
static const struct snd_pcm_ops snd_ad1889_playback_ops = {
.open = snd_ad1889_playback_open,
.close = snd_ad1889_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -582,7 +582,7 @@ static struct snd_pcm_ops snd_ad1889_playback_ops = {
.pointer = snd_ad1889_playback_pointer,
};
static struct snd_pcm_ops snd_ad1889_capture_ops = {
static const struct snd_pcm_ops snd_ad1889_capture_ops = {
.open = snd_ad1889_capture_open,
.close = snd_ad1889_capture_close,
.ioctl = snd_pcm_lib_ioctl,

2
sound/pci/ali5451/ali5451.c
View File

@ -1408,6 +1408,7 @@ snd_ali_playback_pointer(struct snd_pcm_substream *substream)
spin_unlock(&codec->reg_lock);
dev_dbg(codec->card->dev, "playback pointer returned cso=%xh.\n", cso);
cso %= runtime->buffer_size;
return cso;
}
@ -1428,6 +1429,7 @@ static snd_pcm_uframes_t snd_ali_pointer(struct snd_pcm_substream *substream)
cso = inw(ALI_REG(codec, ALI_CSO_ALPHA_FMS + 2));
spin_unlock(&codec->reg_lock);
cso %= runtime->buffer_size;
return cso;
}

4
sound/pci/als300.c
View File

@ -563,7 +563,7 @@ static snd_pcm_uframes_t snd_als300_pointer(struct snd_pcm_substream *substream)
return bytes_to_frames(substream->runtime, current_ptr);
}
static struct snd_pcm_ops snd_als300_playback_ops = {
static const struct snd_pcm_ops snd_als300_playback_ops = {
.open = snd_als300_playback_open,
.close = snd_als300_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -574,7 +574,7 @@ static struct snd_pcm_ops snd_als300_playback_ops = {
.pointer = snd_als300_pointer,
};
static struct snd_pcm_ops snd_als300_capture_ops = {
static const struct snd_pcm_ops snd_als300_capture_ops = {
.open = snd_als300_capture_open,
.close = snd_als300_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/als4000.c
View File

@ -672,7 +672,7 @@ static int snd_als4000_capture_close(struct snd_pcm_substream *substream)
/******************************************************************/
static struct snd_pcm_ops snd_als4000_playback_ops = {
static const struct snd_pcm_ops snd_als4000_playback_ops = {
.open = snd_als4000_playback_open,
.close = snd_als4000_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -683,7 +683,7 @@ static struct snd_pcm_ops snd_als4000_playback_ops = {
.pointer = snd_als4000_playback_pointer
};
static struct snd_pcm_ops snd_als4000_capture_ops = {
static const struct snd_pcm_ops snd_als4000_capture_ops = {
.open = snd_als4000_capture_open,
.close = snd_als4000_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/asihpi/asihpi.c
View File

@ -1138,7 +1138,7 @@ static int snd_card_asihpi_playback_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_card_asihpi_playback_mmap_ops = {
static const struct snd_pcm_ops snd_card_asihpi_playback_mmap_ops = {
.open = snd_card_asihpi_playback_open,
.close = snd_card_asihpi_playback_close,
.ioctl = snd_card_asihpi_playback_ioctl,
@ -1305,7 +1305,7 @@ static int snd_card_asihpi_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_card_asihpi_capture_mmap_ops = {
static const struct snd_pcm_ops snd_card_asihpi_capture_mmap_ops = {
.open = snd_card_asihpi_capture_open,
.close = snd_card_asihpi_capture_close,
.ioctl = snd_card_asihpi_capture_ioctl,

7
sound/pci/asihpi/hpifunc.c
View File

@ -2323,11 +2323,8 @@ u16 hpi_sample_clock_get_source_index(u32 h_control, u16 *pw_source_index)
u16 hpi_sample_clock_query_local_rate(const u32 h_clock, const u32 index,
u32 *prate)
{
u16 err;
err = hpi_control_query(h_clock, HPI_SAMPLECLOCK_LOCAL_SAMPLERATE,
index, 0, prate);
return err;
return hpi_control_query(h_clock, HPI_SAMPLECLOCK_LOCAL_SAMPLERATE,
index, 0, prate);
}
u16 hpi_sample_clock_set_local_rate(u32 h_control, u32 sample_rate)

6
sound/pci/atiixp.c
View File

@ -1148,7 +1148,7 @@ static int snd_atiixp_spdif_close(struct snd_pcm_substream *substream)
}
/* AC97 playback */
static struct snd_pcm_ops snd_atiixp_playback_ops = {
static const struct snd_pcm_ops snd_atiixp_playback_ops = {
.open = snd_atiixp_playback_open,
.close = snd_atiixp_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1160,7 +1160,7 @@ static struct snd_pcm_ops snd_atiixp_playback_ops = {
};
/* AC97 capture */
static struct snd_pcm_ops snd_atiixp_capture_ops = {
static const struct snd_pcm_ops snd_atiixp_capture_ops = {
.open = snd_atiixp_capture_open,
.close = snd_atiixp_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1172,7 +1172,7 @@ static struct snd_pcm_ops snd_atiixp_capture_ops = {
};
/* SPDIF playback */
static struct snd_pcm_ops snd_atiixp_spdif_ops = {
static const struct snd_pcm_ops snd_atiixp_spdif_ops = {
.open = snd_atiixp_spdif_open,
.close = snd_atiixp_spdif_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/atiixp_modem.c
View File

@ -947,7 +947,7 @@ static int snd_atiixp_capture_close(struct snd_pcm_substream *substream)
/* AC97 playback */
static struct snd_pcm_ops snd_atiixp_playback_ops = {
static const struct snd_pcm_ops snd_atiixp_playback_ops = {
.open = snd_atiixp_playback_open,
.close = snd_atiixp_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -959,7 +959,7 @@ static struct snd_pcm_ops snd_atiixp_playback_ops = {
};
/* AC97 capture */
static struct snd_pcm_ops snd_atiixp_capture_ops = {
static const struct snd_pcm_ops snd_atiixp_capture_ops = {
.open = snd_atiixp_capture_open,
.close = snd_atiixp_capture_close,
.ioctl = snd_pcm_lib_ioctl,

2
sound/pci/au88x0/au88x0_core.c
View File

@ -1043,7 +1043,7 @@ static void vortex_fifo_init(vortex_t * vortex)
for (x = NR_ADB - 1; x >= 0; x--) {
hwwrite(vortex->mmio, addr, (FIFO_U0 | FIFO_U1));
if (hwread(vortex->mmio, addr) != (FIFO_U0 | FIFO_U1))
dev_err(vortex->card->dev, "bad adb fifo reset!");
dev_err(vortex->card->dev, "bad adb fifo reset!\n");
vortex_fifo_clearadbdata(vortex, x, FIFO_SIZE);
addr -= 4;
}

4
sound/pci/aw2/aw2-alsa.c
View File

@ -179,7 +179,7 @@ static struct pci_driver aw2_driver = {
module_pci_driver(aw2_driver);
/* operators for playback PCM alsa interface */
static struct snd_pcm_ops snd_aw2_playback_ops = {
static const struct snd_pcm_ops snd_aw2_playback_ops = {
.open = snd_aw2_pcm_playback_open,
.close = snd_aw2_pcm_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -191,7 +191,7 @@ static struct snd_pcm_ops snd_aw2_playback_ops = {
};
/* operators for capture PCM alsa interface */
static struct snd_pcm_ops snd_aw2_capture_ops = {
static const struct snd_pcm_ops snd_aw2_capture_ops = {
.open = snd_aw2_pcm_capture_open,
.close = snd_aw2_pcm_capture_close,
.ioctl = snd_pcm_lib_ioctl,

6
sound/pci/azt3328.c
View File

@ -2090,7 +2090,7 @@ snd_azf3328_pcm_close(struct snd_pcm_substream *substream
/******************************************************************/
static struct snd_pcm_ops snd_azf3328_playback_ops = {
static const struct snd_pcm_ops snd_azf3328_playback_ops = {
.open = snd_azf3328_pcm_playback_open,
.close = snd_azf3328_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -2101,7 +2101,7 @@ static struct snd_pcm_ops snd_azf3328_playback_ops = {
.pointer = snd_azf3328_pcm_pointer
};
static struct snd_pcm_ops snd_azf3328_capture_ops = {
static const struct snd_pcm_ops snd_azf3328_capture_ops = {
.open = snd_azf3328_pcm_capture_open,
.close = snd_azf3328_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -2112,7 +2112,7 @@ static struct snd_pcm_ops snd_azf3328_capture_ops = {
.pointer = snd_azf3328_pcm_pointer
};
static struct snd_pcm_ops snd_azf3328_i2s_out_ops = {
static const struct snd_pcm_ops snd_azf3328_i2s_out_ops = {
.open = snd_azf3328_pcm_i2s_out_open,
.close = snd_azf3328_pcm_close,
.ioctl = snd_pcm_lib_ioctl,

2
sound/pci/bt87x.c
View File

@ -550,7 +550,7 @@ static snd_pcm_uframes_t snd_bt87x_pointer(struct snd_pcm_substream *substream)
return (snd_pcm_uframes_t)bytes_to_frames(runtime, chip->current_line * chip->line_bytes);
}
static struct snd_pcm_ops snd_bt87x_pcm_ops = {
static const struct snd_pcm_ops snd_bt87x_pcm_ops = {
.open = snd_bt87x_pcm_open,
.close = snd_bt87x_close,
.ioctl = snd_pcm_lib_ioctl,

16
sound/pci/ca0106/ca0106_main.c
View File

@ -1109,7 +1109,7 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
}
/* operators */
static struct snd_pcm_ops snd_ca0106_playback_front_ops = {
static const struct snd_pcm_ops snd_ca0106_playback_front_ops = {
.open = snd_ca0106_pcm_open_playback_front,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
@ -1120,7 +1120,7 @@ static struct snd_pcm_ops snd_ca0106_playback_front_ops = {
.pointer = snd_ca0106_pcm_pointer_playback,
};
static struct snd_pcm_ops snd_ca0106_capture_0_ops = {
static const struct snd_pcm_ops snd_ca0106_capture_0_ops = {
.open = snd_ca0106_pcm_open_0_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
@ -1131,7 +1131,7 @@ static struct snd_pcm_ops snd_ca0106_capture_0_ops = {
.pointer = snd_ca0106_pcm_pointer_capture,
};
static struct snd_pcm_ops snd_ca0106_capture_1_ops = {
static const struct snd_pcm_ops snd_ca0106_capture_1_ops = {
.open = snd_ca0106_pcm_open_1_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
@ -1142,7 +1142,7 @@ static struct snd_pcm_ops snd_ca0106_capture_1_ops = {
.pointer = snd_ca0106_pcm_pointer_capture,
};
static struct snd_pcm_ops snd_ca0106_capture_2_ops = {
static const struct snd_pcm_ops snd_ca0106_capture_2_ops = {
.open = snd_ca0106_pcm_open_2_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
@ -1153,7 +1153,7 @@ static struct snd_pcm_ops snd_ca0106_capture_2_ops = {
.pointer = snd_ca0106_pcm_pointer_capture,
};
static struct snd_pcm_ops snd_ca0106_capture_3_ops = {
static const struct snd_pcm_ops snd_ca0106_capture_3_ops = {
.open = snd_ca0106_pcm_open_3_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
@ -1164,7 +1164,7 @@ static struct snd_pcm_ops snd_ca0106_capture_3_ops = {
.pointer = snd_ca0106_pcm_pointer_capture,
};
static struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = {
static const struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = {
.open = snd_ca0106_pcm_open_playback_center_lfe,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
@ -1175,7 +1175,7 @@ static struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = {
.pointer = snd_ca0106_pcm_pointer_playback,
};
static struct snd_pcm_ops snd_ca0106_playback_unknown_ops = {
static const struct snd_pcm_ops snd_ca0106_playback_unknown_ops = {
.open = snd_ca0106_pcm_open_playback_unknown,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
@ -1186,7 +1186,7 @@ static struct snd_pcm_ops snd_ca0106_playback_unknown_ops = {
.pointer = snd_ca0106_pcm_pointer_playback,
};
static struct snd_pcm_ops snd_ca0106_playback_rear_ops = {
static const struct snd_pcm_ops snd_ca0106_playback_rear_ops = {
.open = snd_ca0106_pcm_open_playback_rear,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,

10
sound/pci/cmipci.c
View File

@ -1838,7 +1838,7 @@ static int snd_cmipci_capture_spdif_close(struct snd_pcm_substream *substream)
/*
*/
static struct snd_pcm_ops snd_cmipci_playback_ops = {
static const struct snd_pcm_ops snd_cmipci_playback_ops = {
.open = snd_cmipci_playback_open,
.close = snd_cmipci_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1849,7 +1849,7 @@ static struct snd_pcm_ops snd_cmipci_playback_ops = {
.pointer = snd_cmipci_playback_pointer,
};
static struct snd_pcm_ops snd_cmipci_capture_ops = {
static const struct snd_pcm_ops snd_cmipci_capture_ops = {
.open = snd_cmipci_capture_open,
.close = snd_cmipci_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1860,7 +1860,7 @@ static struct snd_pcm_ops snd_cmipci_capture_ops = {
.pointer = snd_cmipci_capture_pointer,
};
static struct snd_pcm_ops snd_cmipci_playback2_ops = {
static const struct snd_pcm_ops snd_cmipci_playback2_ops = {
.open = snd_cmipci_playback2_open,
.close = snd_cmipci_playback2_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1871,7 +1871,7 @@ static struct snd_pcm_ops snd_cmipci_playback2_ops = {
.pointer = snd_cmipci_capture_pointer, /* channel B */
};
static struct snd_pcm_ops snd_cmipci_playback_spdif_ops = {
static const struct snd_pcm_ops snd_cmipci_playback_spdif_ops = {
.open = snd_cmipci_playback_spdif_open,
.close = snd_cmipci_playback_spdif_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1882,7 +1882,7 @@ static struct snd_pcm_ops snd_cmipci_playback_spdif_ops = {
.pointer = snd_cmipci_playback_pointer,
};
static struct snd_pcm_ops snd_cmipci_capture_spdif_ops = {
static const struct snd_pcm_ops snd_cmipci_capture_spdif_ops = {
.open = snd_cmipci_capture_spdif_open,
.close = snd_cmipci_capture_spdif_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/cs4281.c
View File

@ -951,7 +951,7 @@ static int snd_cs4281_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_cs4281_playback_ops = {
static const struct snd_pcm_ops snd_cs4281_playback_ops = {
.open = snd_cs4281_playback_open,
.close = snd_cs4281_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -962,7 +962,7 @@ static struct snd_pcm_ops snd_cs4281_playback_ops = {
.pointer = snd_cs4281_pointer,
};
static struct snd_pcm_ops snd_cs4281_capture_ops = {
static const struct snd_pcm_ops snd_cs4281_capture_ops = {
.open = snd_cs4281_capture_open,
.close = snd_cs4281_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/cs46xx/cs46xx_lib.c
View File

@ -2460,7 +2460,7 @@ static int cs46xx_detect_codec(struct snd_cs46xx *chip, int codec)
udelay(10);
if (snd_cs46xx_codec_read(chip, AC97_RESET, codec) & 0x8000) {
dev_dbg(chip->card->dev,
"seconadry codec not present\n");
"secondary codec not present\n");
return -ENXIO;
}
}
@ -2503,7 +2503,7 @@ int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device)
chip->nr_ac97_codecs = 1;
#ifdef CONFIG_SND_CS46XX_NEW_DSP
dev_dbg(chip->card->dev, "detecting seconadry codec\n");
dev_dbg(chip->card->dev, "detecting secondary codec\n");
/* try detect a secondary codec */
if (! cs46xx_detect_codec(chip, CS46XX_SECONDARY_CODEC_INDEX))
chip->nr_ac97_codecs = 2;

2
sound/pci/cs46xx/dsp_spos.c
View File

@ -1441,7 +1441,7 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip)
if (chip->nr_ac97_codecs == 2) {
/* create CODEC tasklet for rear Center/LFE output
slot 6 and 9 on seconadry CODEC */
slot 6 and 9 on secondary CODEC */
clfe_codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,"CodecOutSCB_CLFE",0x0030,0x0030,
CLFE_MIXER_SCB_ADDR,
CLFE_CODEC_SCB_ADDR,

4
sound/pci/cs5535audio/cs5535audio_pcm.c
View File

@ -380,7 +380,7 @@ static int snd_cs5535audio_capture_prepare(struct snd_pcm_substream *substream)
substream->runtime->rate);
}
static struct snd_pcm_ops snd_cs5535audio_playback_ops = {
static const struct snd_pcm_ops snd_cs5535audio_playback_ops = {
.open = snd_cs5535audio_playback_open,
.close = snd_cs5535audio_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -391,7 +391,7 @@ static struct snd_pcm_ops snd_cs5535audio_playback_ops = {
.pointer = snd_cs5535audio_pcm_pointer,
};
static struct snd_pcm_ops snd_cs5535audio_capture_ops = {
static const struct snd_pcm_ops snd_cs5535audio_capture_ops = {
.open = snd_cs5535audio_capture_open,
.close = snd_cs5535audio_capture_close,
.ioctl = snd_pcm_lib_ioctl,

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

@ -1623,7 +1623,7 @@ static int atc_resume(struct ct_atc *atc)
}
#endif
static struct ct_atc atc_preset = {
static const struct ct_atc atc_preset = {
.map_audio_buffer = ct_map_audio_buffer,
.unmap_audio_buffer = ct_unmap_audio_buffer,
.pcm_playback_prepare = atc_pcm_playback_prepare,

4
sound/pci/ctxfi/ctpcm.c
View File

@ -372,7 +372,7 @@ ct_pcm_capture_pointer(struct snd_pcm_substream *substream)
}
/* PCM operators for playback */
static struct snd_pcm_ops ct_pcm_playback_ops = {
static const struct snd_pcm_ops ct_pcm_playback_ops = {
.open = ct_pcm_playback_open,
.close = ct_pcm_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -385,7 +385,7 @@ static struct snd_pcm_ops ct_pcm_playback_ops = {
};
/* PCM operators for capture */
static struct snd_pcm_ops ct_pcm_capture_ops = {
static const struct snd_pcm_ops ct_pcm_capture_ops = {
.open = ct_pcm_capture_open,
.close = ct_pcm_capture_close,
.ioctl = snd_pcm_lib_ioctl,

6
sound/pci/ctxfi/ctvmem.c
View File

@ -166,11 +166,7 @@ static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block)
static dma_addr_t
ct_get_ptp_phys(struct ct_vm *vm, int index)
{
dma_addr_t addr;
addr = (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr;
return addr;
return (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr;
}
int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci)

4
sound/pci/emu10k1/emu10k1x.c
View File

@ -548,7 +548,7 @@ snd_emu10k1x_pcm_pointer(struct snd_pcm_substream *substream)
}
/* operators */
static struct snd_pcm_ops snd_emu10k1x_playback_ops = {
static const struct snd_pcm_ops snd_emu10k1x_playback_ops = {
.open = snd_emu10k1x_playback_open,
.close = snd_emu10k1x_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -694,7 +694,7 @@ snd_emu10k1x_pcm_pointer_capture(struct snd_pcm_substream *substream)
return ptr;
}
static struct snd_pcm_ops snd_emu10k1x_capture_ops = {
static const struct snd_pcm_ops snd_emu10k1x_capture_ops = {
.open = snd_emu10k1x_pcm_open_capture,
.close = snd_emu10k1x_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,

12
sound/pci/emu10k1/emupcm.c
View File

@ -1364,7 +1364,7 @@ static int snd_emu10k1_capture_efx_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_emu10k1_playback_ops = {
static const struct snd_pcm_ops snd_emu10k1_playback_ops = {
.open = snd_emu10k1_playback_open,
.close = snd_emu10k1_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1376,7 +1376,7 @@ static struct snd_pcm_ops snd_emu10k1_playback_ops = {
.page = snd_pcm_sgbuf_ops_page,
};
static struct snd_pcm_ops snd_emu10k1_capture_ops = {
static const struct snd_pcm_ops snd_emu10k1_capture_ops = {
.open = snd_emu10k1_capture_open,
.close = snd_emu10k1_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1388,7 +1388,7 @@ static struct snd_pcm_ops snd_emu10k1_capture_ops = {
};
/* EFX playback */
static struct snd_pcm_ops snd_emu10k1_efx_playback_ops = {
static const struct snd_pcm_ops snd_emu10k1_efx_playback_ops = {
.open = snd_emu10k1_efx_playback_open,
.close = snd_emu10k1_efx_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1455,7 +1455,7 @@ int snd_emu10k1_pcm_multi(struct snd_emu10k1 *emu, int device)
}
static struct snd_pcm_ops snd_emu10k1_capture_mic_ops = {
static const struct snd_pcm_ops snd_emu10k1_capture_mic_ops = {
.open = snd_emu10k1_capture_mic_open,
.close = snd_emu10k1_capture_mic_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1550,7 +1550,7 @@ static struct snd_kcontrol_new snd_emu10k1_pcm_efx_voices_mask = {
.put = snd_emu10k1_pcm_efx_voices_mask_put
};
static struct snd_pcm_ops snd_emu10k1_capture_efx_ops = {
static const struct snd_pcm_ops snd_emu10k1_capture_efx_ops = {
.open = snd_emu10k1_capture_efx_open,
.close = snd_emu10k1_capture_efx_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1791,7 +1791,7 @@ static int snd_emu10k1_fx8010_playback_close(struct snd_pcm_substream *substream
return 0;
}
static struct snd_pcm_ops snd_emu10k1_fx8010_playback_ops = {
static const struct snd_pcm_ops snd_emu10k1_fx8010_playback_ops = {
.open = snd_emu10k1_fx8010_playback_open,
.close = snd_emu10k1_fx8010_playback_close,
.ioctl = snd_pcm_lib_ioctl,

20
sound/pci/emu10k1/p16v.c
View File

@ -300,37 +300,29 @@ static int snd_p16v_pcm_open_capture(struct snd_pcm_substream *substream)
static int snd_p16v_pcm_hw_params_playback(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
int result;
result = snd_pcm_lib_malloc_pages(substream,
return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
return result;
}
/* hw_params callback */
static int snd_p16v_pcm_hw_params_capture(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
int result;
result = snd_pcm_lib_malloc_pages(substream,
return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
return result;
}
/* hw_free callback */
static int snd_p16v_pcm_hw_free_playback(struct snd_pcm_substream *substream)
{
int result;
result = snd_pcm_lib_free_pages(substream);
return result;
return snd_pcm_lib_free_pages(substream);
}
/* hw_free callback */
static int snd_p16v_pcm_hw_free_capture(struct snd_pcm_substream *substream)
{
int result;
result = snd_pcm_lib_free_pages(substream);
return result;
return snd_pcm_lib_free_pages(substream);
}
@ -601,7 +593,7 @@ snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream)
}
/* operators */
static struct snd_pcm_ops snd_p16v_playback_front_ops = {
static const struct snd_pcm_ops snd_p16v_playback_front_ops = {
.open = snd_p16v_pcm_open_playback_front,
.close = snd_p16v_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
@ -612,7 +604,7 @@ static struct snd_pcm_ops snd_p16v_playback_front_ops = {
.pointer = snd_p16v_pcm_pointer_playback,
};
static struct snd_pcm_ops snd_p16v_capture_ops = {
static const struct snd_pcm_ops snd_p16v_capture_ops = {
.open = snd_p16v_pcm_open_capture,
.close = snd_p16v_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,

6
sound/pci/ens1370.c
View File

@ -1227,7 +1227,7 @@ static int snd_ensoniq_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_ensoniq_playback1_ops = {
static const struct snd_pcm_ops snd_ensoniq_playback1_ops = {
.open = snd_ensoniq_playback1_open,
.close = snd_ensoniq_playback1_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1238,7 +1238,7 @@ static struct snd_pcm_ops snd_ensoniq_playback1_ops = {
.pointer = snd_ensoniq_playback1_pointer,
};
static struct snd_pcm_ops snd_ensoniq_playback2_ops = {
static const struct snd_pcm_ops snd_ensoniq_playback2_ops = {
.open = snd_ensoniq_playback2_open,
.close = snd_ensoniq_playback2_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1249,7 +1249,7 @@ static struct snd_pcm_ops snd_ensoniq_playback2_ops = {
.pointer = snd_ensoniq_playback2_pointer,
};
static struct snd_pcm_ops snd_ensoniq_capture_ops = {
static const struct snd_pcm_ops snd_ensoniq_capture_ops = {
.open = snd_ensoniq_capture_open,
.close = snd_ensoniq_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/es1938.c
View File

@ -992,7 +992,7 @@ static int snd_es1938_playback_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_es1938_playback_ops = {
static const struct snd_pcm_ops snd_es1938_playback_ops = {
.open = snd_es1938_playback_open,
.close = snd_es1938_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1003,7 +1003,7 @@ static struct snd_pcm_ops snd_es1938_playback_ops = {
.pointer = snd_es1938_playback_pointer,
};
static struct snd_pcm_ops snd_es1938_capture_ops = {
static const struct snd_pcm_ops snd_es1938_capture_ops = {
.open = snd_es1938_capture_open,
.close = snd_es1938_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/es1968.c
View File

@ -1677,7 +1677,7 @@ static int snd_es1968_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_es1968_playback_ops = {
static const struct snd_pcm_ops snd_es1968_playback_ops = {
.open = snd_es1968_playback_open,
.close = snd_es1968_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1688,7 +1688,7 @@ static struct snd_pcm_ops snd_es1968_playback_ops = {
.pointer = snd_es1968_pcm_pointer,
};
static struct snd_pcm_ops snd_es1968_capture_ops = {
static const struct snd_pcm_ops snd_es1968_capture_ops = {
.open = snd_es1968_capture_open,
.close = snd_es1968_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/fm801.c
View File

@ -691,7 +691,7 @@ static int snd_fm801_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_fm801_playback_ops = {
static const struct snd_pcm_ops snd_fm801_playback_ops = {
.open = snd_fm801_playback_open,
.close = snd_fm801_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -702,7 +702,7 @@ static struct snd_pcm_ops snd_fm801_playback_ops = {
.pointer = snd_fm801_playback_pointer,
};
static struct snd_pcm_ops snd_fm801_capture_ops = {
static const struct snd_pcm_ops snd_fm801_capture_ops = {
.open = snd_fm801_capture_open,
.close = snd_fm801_capture_close,
.ioctl = snd_pcm_lib_ioctl,

205
sound/pci/hda/hda_controller.c
View File

@ -27,6 +27,12 @@
#include <linux/module.h>
#include <linux/pm_runtime.h>
#include <linux/slab.h>
#ifdef CONFIG_X86
/* for art-tsc conversion */
#include <asm/tsc.h>
#endif
#include <sound/core.h>
#include <sound/initval.h>
#include "hda_controller.h"
@ -337,12 +343,173 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
azx_get_position(chip, azx_dev));
}
/*
* azx_scale64: Scale base by mult/div while not overflowing sanely
*
* Derived from scale64_check_overflow in kernel/time/timekeeping.c
*
* The tmestamps for a 48Khz stream can overflow after (2^64/10^9)/48K which
* is about 384307 ie ~4.5 days.
*
* This scales the calculation so that overflow will happen but after 2^64 /
* 48000 secs, which is pretty large!
*
* In caln below:
* base may overflow, but since there isn’t any additional division
* performed on base it’s OK
* rem can’t overflow because both are 32-bit values
*/
#ifdef CONFIG_X86
static u64 azx_scale64(u64 base, u32 num, u32 den)
{
u64 rem;
rem = do_div(base, den);
base *= num;
rem *= num;
do_div(rem, den);
return base + rem;
}
static int azx_get_sync_time(ktime_t *device,
struct system_counterval_t *system, void *ctx)
{
struct snd_pcm_substream *substream = ctx;
struct azx_dev *azx_dev = get_azx_dev(substream);
struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
struct azx *chip = apcm->chip;
struct snd_pcm_runtime *runtime;
u64 ll_counter, ll_counter_l, ll_counter_h;
u64 tsc_counter, tsc_counter_l, tsc_counter_h;
u32 wallclk_ctr, wallclk_cycles;
bool direction;
u32 dma_select;
u32 timeout = 200;
u32 retry_count = 0;
runtime = substream->runtime;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
direction = 1;
else
direction = 0;
/* 0th stream tag is not used, so DMA ch 0 is for 1st stream tag */
do {
timeout = 100;
dma_select = (direction << GTSCC_CDMAS_DMA_DIR_SHIFT) |
(azx_dev->core.stream_tag - 1);
snd_hdac_chip_writel(azx_bus(chip), GTSCC, dma_select);
/* Enable the capture */
snd_hdac_chip_updatel(azx_bus(chip), GTSCC, 0, GTSCC_TSCCI_MASK);
while (timeout) {
if (snd_hdac_chip_readl(azx_bus(chip), GTSCC) &
GTSCC_TSCCD_MASK)
break;
timeout--;
}
if (!timeout) {
dev_err(chip->card->dev, "GTSCC capture Timedout!\n");
return -EIO;
}
/* Read wall clock counter */
wallclk_ctr = snd_hdac_chip_readl(azx_bus(chip), WALFCC);
/* Read TSC counter */
tsc_counter_l = snd_hdac_chip_readl(azx_bus(chip), TSCCL);
tsc_counter_h = snd_hdac_chip_readl(azx_bus(chip), TSCCU);
/* Read Link counter */
ll_counter_l = snd_hdac_chip_readl(azx_bus(chip), LLPCL);
ll_counter_h = snd_hdac_chip_readl(azx_bus(chip), LLPCU);
/* Ack: registers read done */
snd_hdac_chip_writel(azx_bus(chip), GTSCC, GTSCC_TSCCD_SHIFT);
tsc_counter = (tsc_counter_h << TSCCU_CCU_SHIFT) |
tsc_counter_l;
ll_counter = (ll_counter_h << LLPC_CCU_SHIFT) | ll_counter_l;
wallclk_cycles = wallclk_ctr & WALFCC_CIF_MASK;
/*
* An error occurs near frame "rollover". The clocks in
* frame value indicates whether this error may have
* occurred. Here we use the value of 10 i.e.,
* HDA_MAX_CYCLE_OFFSET
*/
if (wallclk_cycles < HDA_MAX_CYCLE_VALUE - HDA_MAX_CYCLE_OFFSET
&& wallclk_cycles > HDA_MAX_CYCLE_OFFSET)
break;
/*
* Sleep before we read again, else we may again get
* value near to MAX_CYCLE. Try to sleep for different
* amount of time so we dont hit the same number again
*/
udelay(retry_count++);
} while (retry_count != HDA_MAX_CYCLE_READ_RETRY);
if (retry_count == HDA_MAX_CYCLE_READ_RETRY) {
dev_err_ratelimited(chip->card->dev,
"Error in WALFCC cycle count\n");
return -EIO;
}
*device = ns_to_ktime(azx_scale64(ll_counter,
NSEC_PER_SEC, runtime->rate));
*device = ktime_add_ns(*device, (wallclk_cycles * NSEC_PER_SEC) /
((HDA_MAX_CYCLE_VALUE + 1) * runtime->rate));
*system = convert_art_to_tsc(tsc_counter);
return 0;
}
#else
static int azx_get_sync_time(ktime_t *device,
struct system_counterval_t *system, void *ctx)
{
return -ENXIO;
}
#endif
static int azx_get_crosststamp(struct snd_pcm_substream *substream,
struct system_device_crosststamp *xtstamp)
{
return get_device_system_crosststamp(azx_get_sync_time,
substream, NULL, xtstamp);
}
static inline bool is_link_time_supported(struct snd_pcm_runtime *runtime,
struct snd_pcm_audio_tstamp_config *ts)
{
if (runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME)
if (ts->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)
return true;
return false;
}
static int azx_get_time_info(struct snd_pcm_substream *substream,
struct timespec *system_ts, struct timespec *audio_ts,
struct snd_pcm_audio_tstamp_config *audio_tstamp_config,
struct snd_pcm_audio_tstamp_report *audio_tstamp_report)
{
struct azx_dev *azx_dev = get_azx_dev(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct system_device_crosststamp xtstamp;
int ret;
u64 nsec;
if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) &&
@ -361,8 +528,37 @@ static int azx_get_time_info(struct snd_pcm_substream *substream,
audio_tstamp_report->accuracy_report = 1; /* rest of structure is valid */
audio_tstamp_report->accuracy = 42; /* 24 MHz WallClock == 42ns resolution */
} else
} else if (is_link_time_supported(runtime, audio_tstamp_config)) {
ret = azx_get_crosststamp(substream, &xtstamp);
if (ret)
return ret;
switch (runtime->tstamp_type) {
case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC:
return -EINVAL;
case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW:
*system_ts = ktime_to_timespec(xtstamp.sys_monoraw);
break;
default:
*system_ts = ktime_to_timespec(xtstamp.sys_realtime);
break;
}
*audio_ts = ktime_to_timespec(xtstamp.device);
audio_tstamp_report->actual_type =
SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED;
audio_tstamp_report->accuracy_report = 1;
/* 24 MHz WallClock == 42ns resolution */
audio_tstamp_report->accuracy = 42;
} else {
audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT;
}
return 0;
}
@ -412,6 +608,11 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
goto unlock;
}
runtime->private_data = azx_dev;
if (chip->gts_present)
azx_pcm_hw.info = azx_pcm_hw.info |
SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
runtime->hw = azx_pcm_hw;
runtime->hw.channels_min = hinfo->channels_min;
runtime->hw.channels_max = hinfo->channels_max;
@ -495,7 +696,7 @@ static int azx_pcm_mmap(struct snd_pcm_substream *substream,
return snd_pcm_lib_default_mmap(substream, area);
}
static struct snd_pcm_ops azx_pcm_ops = {
static const struct snd_pcm_ops azx_pcm_ops = {
.open = azx_pcm_open,
.close = azx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,

3
sound/pci/hda/hda_controller.h
View File

@ -159,6 +159,9 @@ struct azx {
unsigned int region_requested:1;
unsigned int disabled:1; /* disabled by vga_switcheroo */
/* GTS present */
unsigned int gts_present:1;
#ifdef CONFIG_SND_HDA_DSP_LOADER
struct azx_dev saved_azx_dev;
#endif

17
sound/pci/hda/hda_intel.c
View File

@ -54,6 +54,7 @@
/* for snoop control */
#include <asm/pgtable.h>
#include <asm/cacheflush.h>
#include <asm/cpufeature.h>
#endif
#include <sound/core.h>
#include <sound/initval.h>
@ -1663,6 +1664,22 @@ static int azx_first_init(struct azx *chip)
return -ENXIO;
}
if (IS_SKL_PLUS(pci))
snd_hdac_bus_parse_capabilities(bus);
/*
* Some Intel CPUs has always running timer (ART) feature and
* controller may have Global time sync reporting capability, so
* check both of these before declaring synchronized time reporting
* capability SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME
*/
chip->gts_present = false;
#ifdef CONFIG_X86
if (bus->ppcap && boot_cpu_has(X86_FEATURE_ART))
chip->gts_present = true;
#endif
if (chip->msi) {
if (chip->driver_caps & AZX_DCAPS_NO_MSI64) {
dev_dbg(card->dev, "Disabling 64bit MSI\n");

10
sound/pci/hda/patch_ca0132.c
View File

@ -4018,7 +4018,7 @@ static int ca0132_build_controls(struct hda_codec *codec)
/*
* PCM
*/
static struct hda_pcm_stream ca0132_pcm_analog_playback = {
static const struct hda_pcm_stream ca0132_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 6,
@ -4029,7 +4029,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
},
};
static struct hda_pcm_stream ca0132_pcm_analog_capture = {
static const struct hda_pcm_stream ca0132_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
@ -4040,7 +4040,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
},
};
static struct hda_pcm_stream ca0132_pcm_digital_playback = {
static const struct hda_pcm_stream ca0132_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
@ -4052,7 +4052,7 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = {
},
};
static struct hda_pcm_stream ca0132_pcm_digital_capture = {
static const struct hda_pcm_stream ca0132_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
@ -4614,7 +4614,7 @@ static void ca0132_free(struct hda_codec *codec)
kfree(codec->spec);
}
static struct hda_codec_ops ca0132_patch_ops = {
static const struct hda_codec_ops ca0132_patch_ops = {
.build_controls = ca0132_build_controls,
.build_pcms = ca0132_build_pcms,
.init = ca0132_init,

10
sound/pci/hda/patch_conexant.c
View File

@ -261,6 +261,7 @@ enum {
CXT_FIXUP_HP_530,
CXT_FIXUP_CAP_MIX_AMP_5047,
CXT_FIXUP_MUTE_LED_EAPD,
CXT_FIXUP_HP_SPECTRE,
};
/* for hda_fixup_thinkpad_acpi() */
@ -765,6 +766,14 @@ static const struct hda_fixup cxt_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = cxt_fixup_mute_led_eapd,
},
[CXT_FIXUP_HP_SPECTRE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
/* enable NID 0x1d for the speaker on top */
{ 0x1d, 0x91170111 },
{ }
}
},
};
static const struct snd_pci_quirk cxt5045_fixups[] = {
@ -814,6 +823,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC),
SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC),
SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),

19
sound/pci/hda/patch_realtek.c
View File

@ -5264,6 +5264,8 @@ static const struct hda_fixup alc269_fixups[] = {
[ALC269_FIXUP_THINKPAD_ACPI] = {
.type = HDA_FIXUP_FUNC,
.v.func = hda_fixup_thinkpad_acpi,
.chained = true,
.chain_id = ALC269_FIXUP_SKU_IGNORE,
},
[ALC269_FIXUP_DMIC_THINKPAD_ACPI] = {
.type = HDA_FIXUP_FUNC,
@ -5806,6 +5808,13 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{0x14, 0x90170110}, \
{0x15, 0x0221401f}
#define ALC295_STANDARD_PINS \
{0x12, 0xb7a60130}, \
{0x14, 0x90170110}, \
{0x17, 0x21014020}, \
{0x18, 0x21a19030}, \
{0x21, 0x04211020}
#define ALC298_STANDARD_PINS \
{0x12, 0x90a60130}, \
{0x21, 0x03211020}
@ -5845,6 +5854,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x12, 0x90a60160},
{0x14, 0x90170120},
{0x21, 0x02211030}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x14, 0x90170110},
{0x1b, 0x02011020},
{0x21, 0x0221101f}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x14, 0x90170130},
{0x1b, 0x01014020},
@ -5910,6 +5923,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x12, 0x90a60180},
{0x14, 0x90170120},
{0x21, 0x02211030}),
SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x12, 0xb7a60130},
{0x14, 0x90170110},
{0x21, 0x02211020}),
SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC256_STANDARD_PINS),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
@ -6021,6 +6038,8 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC292_STANDARD_PINS,
{0x13, 0x90a60140}),
SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC295_STANDARD_PINS),
SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC298_STANDARD_PINS,
{0x17, 0x90170110}),

10
sound/pci/ice1712/ice1712.c
View File

@ -847,7 +847,7 @@ static int snd_ice1712_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_ice1712_playback_ops = {
static const struct snd_pcm_ops snd_ice1712_playback_ops = {
.open = snd_ice1712_playback_open,
.close = snd_ice1712_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -858,7 +858,7 @@ static struct snd_pcm_ops snd_ice1712_playback_ops = {
.pointer = snd_ice1712_playback_pointer,
};
static struct snd_pcm_ops snd_ice1712_playback_ds_ops = {
static const struct snd_pcm_ops snd_ice1712_playback_ds_ops = {
.open = snd_ice1712_playback_ds_open,
.close = snd_ice1712_playback_ds_close,
.ioctl = snd_pcm_lib_ioctl,
@ -869,7 +869,7 @@ static struct snd_pcm_ops snd_ice1712_playback_ds_ops = {
.pointer = snd_ice1712_playback_ds_pointer,
};
static struct snd_pcm_ops snd_ice1712_capture_ops = {
static const struct snd_pcm_ops snd_ice1712_capture_ops = {
.open = snd_ice1712_capture_open,
.close = snd_ice1712_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1228,7 +1228,7 @@ static int snd_ice1712_capture_pro_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_ice1712_playback_pro_ops = {
static const struct snd_pcm_ops snd_ice1712_playback_pro_ops = {
.open = snd_ice1712_playback_pro_open,
.close = snd_ice1712_playback_pro_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1239,7 +1239,7 @@ static struct snd_pcm_ops snd_ice1712_playback_pro_ops = {
.pointer = snd_ice1712_playback_pro_pointer,
};
static struct snd_pcm_ops snd_ice1712_capture_pro_ops = {
static const struct snd_pcm_ops snd_ice1712_capture_pro_ops = {
.open = snd_ice1712_capture_pro_open,
.close = snd_ice1712_capture_pro_close,
.ioctl = snd_pcm_lib_ioctl,

14
sound/pci/ice1712/ice1724.c
View File

@ -620,9 +620,7 @@ static const unsigned int stdclock_rate_list[16] = {
static unsigned int stdclock_get_rate(struct snd_ice1712 *ice)
{
unsigned int rate;
rate = stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15];
return rate;
return stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15];
}
static void stdclock_set_rate(struct snd_ice1712 *ice, unsigned int rate)
@ -1113,7 +1111,7 @@ static int snd_vt1724_capture_pro_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_vt1724_playback_pro_ops = {
static const struct snd_pcm_ops snd_vt1724_playback_pro_ops = {
.open = snd_vt1724_playback_pro_open,
.close = snd_vt1724_playback_pro_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1124,7 +1122,7 @@ static struct snd_pcm_ops snd_vt1724_playback_pro_ops = {
.pointer = snd_vt1724_playback_pro_pointer,
};
static struct snd_pcm_ops snd_vt1724_capture_pro_ops = {
static const struct snd_pcm_ops snd_vt1724_capture_pro_ops = {
.open = snd_vt1724_capture_pro_open,
.close = snd_vt1724_capture_pro_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1292,7 +1290,7 @@ static int snd_vt1724_capture_spdif_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_vt1724_playback_spdif_ops = {
static const struct snd_pcm_ops snd_vt1724_playback_spdif_ops = {
.open = snd_vt1724_playback_spdif_open,
.close = snd_vt1724_playback_spdif_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1303,7 +1301,7 @@ static struct snd_pcm_ops snd_vt1724_playback_spdif_ops = {
.pointer = snd_vt1724_pcm_pointer,
};
static struct snd_pcm_ops snd_vt1724_capture_spdif_ops = {
static const struct snd_pcm_ops snd_vt1724_capture_spdif_ops = {
.open = snd_vt1724_capture_spdif_open,
.close = snd_vt1724_capture_spdif_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1437,7 +1435,7 @@ static int snd_vt1724_playback_indep_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_vt1724_playback_indep_ops = {
static const struct snd_pcm_ops snd_vt1724_playback_indep_ops = {
.open = snd_vt1724_playback_indep_open,
.close = snd_vt1724_playback_indep_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/korg1212/korg1212.c
View File

@ -1681,7 +1681,7 @@ static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream,
return snd_korg1212_copy_to(korg1212, dst, pos, count, 0, korg1212->channels * 2);
}
static struct snd_pcm_ops snd_korg1212_playback_ops = {
static const struct snd_pcm_ops snd_korg1212_playback_ops = {
.open = snd_korg1212_playback_open,
.close = snd_korg1212_playback_close,
.ioctl = snd_korg1212_ioctl,
@ -1693,7 +1693,7 @@ static struct snd_pcm_ops snd_korg1212_playback_ops = {
.silence = snd_korg1212_playback_silence,
};
static struct snd_pcm_ops snd_korg1212_capture_ops = {
static const struct snd_pcm_ops snd_korg1212_capture_ops = {
.open = snd_korg1212_capture_open,
.close = snd_korg1212_capture_close,
.ioctl = snd_korg1212_ioctl,

2
sound/pci/lola/lola_pcm.c
View File

@ -586,7 +586,7 @@ void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits)
}
}
static struct snd_pcm_ops lola_pcm_ops = {
static const struct snd_pcm_ops lola_pcm_ops = {
.open = lola_pcm_open,
.close = lola_pcm_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/lx6464es/lx6464es.c
View File

@ -804,7 +804,7 @@ mac_ready:
return err;
}
static struct snd_pcm_ops lx_ops_playback = {
static const struct snd_pcm_ops lx_ops_playback = {
.open = lx_pcm_open,
.close = lx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -815,7 +815,7 @@ static struct snd_pcm_ops lx_ops_playback = {
.pointer = lx_pcm_stream_pointer,
};
static struct snd_pcm_ops lx_ops_capture = {
static const struct snd_pcm_ops lx_ops_capture = {
.open = lx_pcm_open,
.close = lx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/maestro3.c
View File

@ -1834,7 +1834,7 @@ snd_m3_capture_close(struct snd_pcm_substream *subs)
* create pcm instance
*/
static struct snd_pcm_ops snd_m3_playback_ops = {
static const struct snd_pcm_ops snd_m3_playback_ops = {
.open = snd_m3_playback_open,
.close = snd_m3_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1845,7 +1845,7 @@ static struct snd_pcm_ops snd_m3_playback_ops = {
.pointer = snd_m3_pcm_pointer,
};
static struct snd_pcm_ops snd_m3_capture_ops = {
static const struct snd_pcm_ops snd_m3_capture_ops = {
.open = snd_m3_capture_open,
.close = snd_m3_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/mixart/mixart.c
View File

@ -922,7 +922,7 @@ static snd_pcm_uframes_t snd_mixart_stream_pointer(struct snd_pcm_substream *sub
static struct snd_pcm_ops snd_mixart_playback_ops = {
static const struct snd_pcm_ops snd_mixart_playback_ops = {
.open = snd_mixart_playback_open,
.close = snd_mixart_close,
.ioctl = snd_pcm_lib_ioctl,
@ -933,7 +933,7 @@ static struct snd_pcm_ops snd_mixart_playback_ops = {
.pointer = snd_mixart_stream_pointer,
};
static struct snd_pcm_ops snd_mixart_capture_ops = {
static const struct snd_pcm_ops snd_mixart_capture_ops = {
.open = snd_mixart_capture_open,
.close = snd_mixart_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/nm256/nm256.c
View File

@ -902,7 +902,7 @@ snd_nm256_capture_close(struct snd_pcm_substream *substream)
/*
* create a pcm instance
*/
static struct snd_pcm_ops snd_nm256_playback_ops = {
static const struct snd_pcm_ops snd_nm256_playback_ops = {
.open = snd_nm256_playback_open,
.close = snd_nm256_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -917,7 +917,7 @@ static struct snd_pcm_ops snd_nm256_playback_ops = {
.mmap = snd_pcm_lib_mmap_iomem,
};
static struct snd_pcm_ops snd_nm256_capture_ops = {
static const struct snd_pcm_ops snd_nm256_capture_ops = {
.open = snd_nm256_capture_open,
.close = snd_nm256_capture_close,
.ioctl = snd_pcm_lib_ioctl,

12
sound/pci/oxygen/oxygen_pcm.c
View File

@ -631,7 +631,7 @@ static snd_pcm_uframes_t oxygen_pointer(struct snd_pcm_substream *substream)
return bytes_to_frames(runtime, curr_addr - (u32)runtime->dma_addr);
}
static struct snd_pcm_ops oxygen_rec_a_ops = {
static const struct snd_pcm_ops oxygen_rec_a_ops = {
.open = oxygen_rec_a_open,
.close = oxygen_close,
.ioctl = snd_pcm_lib_ioctl,
@ -642,7 +642,7 @@ static struct snd_pcm_ops oxygen_rec_a_ops = {
.pointer = oxygen_pointer,
};
static struct snd_pcm_ops oxygen_rec_b_ops = {
static const struct snd_pcm_ops oxygen_rec_b_ops = {
.open = oxygen_rec_b_open,
.close = oxygen_close,
.ioctl = snd_pcm_lib_ioctl,
@ -653,7 +653,7 @@ static struct snd_pcm_ops oxygen_rec_b_ops = {
.pointer = oxygen_pointer,
};
static struct snd_pcm_ops oxygen_rec_c_ops = {
static const struct snd_pcm_ops oxygen_rec_c_ops = {
.open = oxygen_rec_c_open,
.close = oxygen_close,
.ioctl = snd_pcm_lib_ioctl,
@ -664,7 +664,7 @@ static struct snd_pcm_ops oxygen_rec_c_ops = {
.pointer = oxygen_pointer,
};
static struct snd_pcm_ops oxygen_spdif_ops = {
static const struct snd_pcm_ops oxygen_spdif_ops = {
.open = oxygen_spdif_open,
.close = oxygen_close,
.ioctl = snd_pcm_lib_ioctl,
@ -675,7 +675,7 @@ static struct snd_pcm_ops oxygen_spdif_ops = {
.pointer = oxygen_pointer,
};
static struct snd_pcm_ops oxygen_multich_ops = {
static const struct snd_pcm_ops oxygen_multich_ops = {
.open = oxygen_multich_open,
.close = oxygen_close,
.ioctl = snd_pcm_lib_ioctl,
@ -686,7 +686,7 @@ static struct snd_pcm_ops oxygen_multich_ops = {
.pointer = oxygen_pointer,
};
static struct snd_pcm_ops oxygen_ac97_ops = {
static const struct snd_pcm_ops oxygen_ac97_ops = {
.open = oxygen_ac97_open,
.close = oxygen_close,
.ioctl = snd_pcm_lib_ioctl,

2
sound/pci/pcxhr/pcxhr.c
View File

@ -1146,7 +1146,7 @@ static snd_pcm_uframes_t pcxhr_stream_pointer(struct snd_pcm_substream *subs)
}
static struct snd_pcm_ops pcxhr_ops = {
static const struct snd_pcm_ops pcxhr_ops = {
.open = pcxhr_open,
.close = pcxhr_close,
.ioctl = snd_pcm_lib_ioctl,

6
sound/pci/riptide/riptide.c
View File

@ -644,7 +644,7 @@ static struct lbuspath lbus_play_paths[] = {
.mono = lbus_play_mono3,
},
};
static struct lbuspath lbus_rec_path = {
static const struct lbuspath lbus_rec_path = {
.noconv = lbus_rec_noconv1,
.stereo = lbus_rec_stereo1,
.mono = lbus_rec_mono1,
@ -1669,7 +1669,7 @@ static int snd_riptide_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_riptide_playback_ops = {
static const struct snd_pcm_ops snd_riptide_playback_ops = {
.open = snd_riptide_playback_open,
.close = snd_riptide_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1680,7 +1680,7 @@ static struct snd_pcm_ops snd_riptide_playback_ops = {
.trigger = snd_riptide_trigger,
.pointer = snd_riptide_pointer,
};
static struct snd_pcm_ops snd_riptide_capture_ops = {
static const struct snd_pcm_ops snd_riptide_capture_ops = {
.open = snd_riptide_capture_open,
.close = snd_riptide_capture_close,
.ioctl = snd_pcm_lib_ioctl,

16
sound/pci/rme32.c
View File

@ -1196,7 +1196,7 @@ snd_rme32_capture_fd_pointer(struct snd_pcm_substream *substream)
}
/* for halfduplex mode */
static struct snd_pcm_ops snd_rme32_playback_spdif_ops = {
static const struct snd_pcm_ops snd_rme32_playback_spdif_ops = {
.open = snd_rme32_playback_spdif_open,
.close = snd_rme32_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1210,7 +1210,7 @@ static struct snd_pcm_ops snd_rme32_playback_spdif_ops = {
.mmap = snd_pcm_lib_mmap_iomem,
};
static struct snd_pcm_ops snd_rme32_capture_spdif_ops = {
static const struct snd_pcm_ops snd_rme32_capture_spdif_ops = {
.open = snd_rme32_capture_spdif_open,
.close = snd_rme32_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1223,7 +1223,7 @@ static struct snd_pcm_ops snd_rme32_capture_spdif_ops = {
.mmap = snd_pcm_lib_mmap_iomem,
};
static struct snd_pcm_ops snd_rme32_playback_adat_ops = {
static const struct snd_pcm_ops snd_rme32_playback_adat_ops = {
.open = snd_rme32_playback_adat_open,
.close = snd_rme32_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1236,7 +1236,7 @@ static struct snd_pcm_ops snd_rme32_playback_adat_ops = {
.mmap = snd_pcm_lib_mmap_iomem,
};
static struct snd_pcm_ops snd_rme32_capture_adat_ops = {
static const struct snd_pcm_ops snd_rme32_capture_adat_ops = {
.open = snd_rme32_capture_adat_open,
.close = snd_rme32_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1249,7 +1249,7 @@ static struct snd_pcm_ops snd_rme32_capture_adat_ops = {
};
/* for fullduplex mode */
static struct snd_pcm_ops snd_rme32_playback_spdif_fd_ops = {
static const struct snd_pcm_ops snd_rme32_playback_spdif_fd_ops = {
.open = snd_rme32_playback_spdif_open,
.close = snd_rme32_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1261,7 +1261,7 @@ static struct snd_pcm_ops snd_rme32_playback_spdif_fd_ops = {
.ack = snd_rme32_playback_fd_ack,
};
static struct snd_pcm_ops snd_rme32_capture_spdif_fd_ops = {
static const struct snd_pcm_ops snd_rme32_capture_spdif_fd_ops = {
.open = snd_rme32_capture_spdif_open,
.close = snd_rme32_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1273,7 +1273,7 @@ static struct snd_pcm_ops snd_rme32_capture_spdif_fd_ops = {
.ack = snd_rme32_capture_fd_ack,
};
static struct snd_pcm_ops snd_rme32_playback_adat_fd_ops = {
static const struct snd_pcm_ops snd_rme32_playback_adat_fd_ops = {
.open = snd_rme32_playback_adat_open,
.close = snd_rme32_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1284,7 +1284,7 @@ static struct snd_pcm_ops snd_rme32_playback_adat_fd_ops = {
.ack = snd_rme32_playback_fd_ack,
};
static struct snd_pcm_ops snd_rme32_capture_adat_fd_ops = {
static const struct snd_pcm_ops snd_rme32_capture_adat_fd_ops = {
.open = snd_rme32_capture_adat_open,
.close = snd_rme32_capture_close,
.ioctl = snd_pcm_lib_ioctl,

8
sound/pci/rme96.c
View File

@ -1505,7 +1505,7 @@ snd_rme96_capture_pointer(struct snd_pcm_substream *substream)
return snd_rme96_capture_ptr(rme96);
}
static struct snd_pcm_ops snd_rme96_playback_spdif_ops = {
static const struct snd_pcm_ops snd_rme96_playback_spdif_ops = {
.open = snd_rme96_playback_spdif_open,
.close = snd_rme96_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1518,7 +1518,7 @@ static struct snd_pcm_ops snd_rme96_playback_spdif_ops = {
.mmap = snd_pcm_lib_mmap_iomem,
};
static struct snd_pcm_ops snd_rme96_capture_spdif_ops = {
static const struct snd_pcm_ops snd_rme96_capture_spdif_ops = {
.open = snd_rme96_capture_spdif_open,
.close = snd_rme96_capture_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1530,7 +1530,7 @@ static struct snd_pcm_ops snd_rme96_capture_spdif_ops = {
.mmap = snd_pcm_lib_mmap_iomem,
};
static struct snd_pcm_ops snd_rme96_playback_adat_ops = {
static const struct snd_pcm_ops snd_rme96_playback_adat_ops = {
.open = snd_rme96_playback_adat_open,
.close = snd_rme96_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1543,7 +1543,7 @@ static struct snd_pcm_ops snd_rme96_playback_adat_ops = {
.mmap = snd_pcm_lib_mmap_iomem,
};
static struct snd_pcm_ops snd_rme96_capture_adat_ops = {
static const struct snd_pcm_ops snd_rme96_capture_adat_ops = {
.open = snd_rme96_capture_adat_open,
.close = snd_rme96_capture_close,
.ioctl = snd_pcm_lib_ioctl,

4
sound/pci/rme9652/hdsp.c
View File

@ -4861,7 +4861,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
return 0;
}
static struct snd_pcm_ops snd_hdsp_playback_ops = {
static const struct snd_pcm_ops snd_hdsp_playback_ops = {
.open = snd_hdsp_playback_open,
.close = snd_hdsp_playback_release,
.ioctl = snd_hdsp_ioctl,
@ -4873,7 +4873,7 @@ static struct snd_pcm_ops snd_hdsp_playback_ops = {
.silence = snd_hdsp_hw_silence,
};
static struct snd_pcm_ops snd_hdsp_capture_ops = {
static const struct snd_pcm_ops snd_hdsp_capture_ops = {
.open = snd_hdsp_capture_open,
.close = snd_hdsp_capture_release,
.ioctl = snd_hdsp_ioctl,

4
sound/pci/rme9652/hdspm.c
View File

@ -1666,7 +1666,7 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally)
HDSPM_AUTOSYNC_FROM_NONE) {
dev_warn(hdspm->card->dev,
"Detected no Externel Sync\n");
"Detected no External Sync\n");
not_set = 1;
} else if (rate != external_freq) {
@ -6361,7 +6361,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
return 0;
}
static struct snd_pcm_ops snd_hdspm_ops = {
static const struct snd_pcm_ops snd_hdspm_ops = {
.open = snd_hdspm_open,
.close = snd_hdspm_release,
.ioctl = snd_hdspm_ioctl,

4
sound/pci/rme9652/rme9652.c
View File

@ -2368,7 +2368,7 @@ static int snd_rme9652_capture_release(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_rme9652_playback_ops = {
static const struct snd_pcm_ops snd_rme9652_playback_ops = {
.open = snd_rme9652_playback_open,
.close = snd_rme9652_playback_release,
.ioctl = snd_rme9652_ioctl,
@ -2380,7 +2380,7 @@ static struct snd_pcm_ops snd_rme9652_playback_ops = {
.silence = snd_rme9652_hw_silence,
};
static struct snd_pcm_ops snd_rme9652_capture_ops = {
static const struct snd_pcm_ops snd_rme9652_capture_ops = {
.open = snd_rme9652_capture_open,
.close = snd_rme9652_capture_release,
.ioctl = snd_rme9652_ioctl,

4
sound/pci/sonicvibes.c
View File

@ -857,7 +857,7 @@ static int snd_sonicvibes_capture_close(struct snd_pcm_substream *substream)
return 0;
}
static struct snd_pcm_ops snd_sonicvibes_playback_ops = {
static const struct snd_pcm_ops snd_sonicvibes_playback_ops = {
.open = snd_sonicvibes_playback_open,
.close = snd_sonicvibes_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -868,7 +868,7 @@ static struct snd_pcm_ops snd_sonicvibes_playback_ops = {
.pointer = snd_sonicvibes_playback_pointer,
};
static struct snd_pcm_ops snd_sonicvibes_capture_ops = {
static const struct snd_pcm_ops snd_sonicvibes_capture_ops = {
.open = snd_sonicvibes_capture_open,
.close = snd_sonicvibes_capture_close,
.ioctl = snd_pcm_lib_ioctl,

12
sound/pci/trident/trident_main.c
View File

@ -2070,7 +2070,7 @@ static int snd_trident_foldback_close(struct snd_pcm_substream *substream)
PCM operations
---------------------------------------------------------------------------*/
static struct snd_pcm_ops snd_trident_playback_ops = {
static const struct snd_pcm_ops snd_trident_playback_ops = {
.open = snd_trident_playback_open,
.close = snd_trident_playback_close,
.ioctl = snd_trident_ioctl,
@ -2081,7 +2081,7 @@ static struct snd_pcm_ops snd_trident_playback_ops = {
.pointer = snd_trident_playback_pointer,
};
static struct snd_pcm_ops snd_trident_nx_playback_ops = {
static const struct snd_pcm_ops snd_trident_nx_playback_ops = {
.open = snd_trident_playback_open,
.close = snd_trident_playback_close,
.ioctl = snd_trident_ioctl,
@ -2115,7 +2115,7 @@ static struct snd_pcm_ops snd_trident_si7018_capture_ops = {
.pointer = snd_trident_playback_pointer,
};
static struct snd_pcm_ops snd_trident_foldback_ops = {
static const struct snd_pcm_ops snd_trident_foldback_ops = {
.open = snd_trident_foldback_open,
.close = snd_trident_foldback_close,
.ioctl = snd_trident_ioctl,
@ -2126,7 +2126,7 @@ static struct snd_pcm_ops snd_trident_foldback_ops = {
.pointer = snd_trident_playback_pointer,
};
static struct snd_pcm_ops snd_trident_nx_foldback_ops = {
static const struct snd_pcm_ops snd_trident_nx_foldback_ops = {
.open = snd_trident_foldback_open,
.close = snd_trident_foldback_close,
.ioctl = snd_trident_ioctl,
@ -2138,7 +2138,7 @@ static struct snd_pcm_ops snd_trident_nx_foldback_ops = {
.page = snd_pcm_sgbuf_ops_page,
};
static struct snd_pcm_ops snd_trident_spdif_ops = {
static const struct snd_pcm_ops snd_trident_spdif_ops = {
.open = snd_trident_spdif_open,
.close = snd_trident_spdif_close,
.ioctl = snd_trident_ioctl,
@ -2149,7 +2149,7 @@ static struct snd_pcm_ops snd_trident_spdif_ops = {
.pointer = snd_trident_spdif_pointer,
};
static struct snd_pcm_ops snd_trident_spdif_7018_ops = {
static const struct snd_pcm_ops snd_trident_spdif_7018_ops = {
.open = snd_trident_spdif_open,
.close = snd_trident_spdif_close,
.ioctl = snd_trident_ioctl,

10
sound/pci/via82xx.c
View File

@ -1366,7 +1366,7 @@ static int snd_via8233_playback_close(struct snd_pcm_substream *substream)
/* via686 playback callbacks */
static struct snd_pcm_ops snd_via686_playback_ops = {
static const struct snd_pcm_ops snd_via686_playback_ops = {
.open = snd_via686_playback_open,
.close = snd_via82xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1379,7 +1379,7 @@ static struct snd_pcm_ops snd_via686_playback_ops = {
};
/* via686 capture callbacks */
static struct snd_pcm_ops snd_via686_capture_ops = {
static const struct snd_pcm_ops snd_via686_capture_ops = {
.open = snd_via82xx_capture_open,
.close = snd_via82xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1392,7 +1392,7 @@ static struct snd_pcm_ops snd_via686_capture_ops = {
};
/* via823x DSX playback callbacks */
static struct snd_pcm_ops snd_via8233_playback_ops = {
static const struct snd_pcm_ops snd_via8233_playback_ops = {
.open = snd_via8233_playback_open,
.close = snd_via8233_playback_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1405,7 +1405,7 @@ static struct snd_pcm_ops snd_via8233_playback_ops = {
};
/* via823x multi-channel playback callbacks */
static struct snd_pcm_ops snd_via8233_multi_ops = {
static const struct snd_pcm_ops snd_via8233_multi_ops = {
.open = snd_via8233_multi_open,
.close = snd_via82xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@ -1418,7 +1418,7 @@ static struct snd_pcm_ops snd_via8233_multi_ops = {
};
/* via823x capture callbacks */
static struct snd_pcm_ops snd_via8233_capture_ops = {
static const struct snd_pcm_ops snd_via8233_capture_ops = {
.open = snd_via82xx_capture_open,
.close = snd_via82xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,

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