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Merge branch 'for-next' into for-linus
This commit is contained in:
Takashi Iwai
@@ -112,6 +112,8 @@
|
||||
!Esound/soc/soc-devres.c
|
||||
!Esound/soc/soc-io.c
|
||||
!Esound/soc/soc-pcm.c
|
||||
!Esound/soc/soc-ops.c
|
||||
!Esound/soc/soc-compress.c
|
||||
</sect1>
|
||||
<sect1><title>ASoC DAPM API</title>
|
||||
!Esound/soc/soc-dapm.c
|
||||
|
||||
@@ -2181,10 +2181,6 @@ struct _snd_pcm_runtime {
|
||||
struct snd_pcm_hardware hw;
|
||||
struct snd_pcm_hw_constraints hw_constraints;
|
||||
|
||||
/* -- interrupt callbacks -- */
|
||||
void (*transfer_ack_begin)(struct snd_pcm_substream *substream);
|
||||
void (*transfer_ack_end)(struct snd_pcm_substream *substream);
|
||||
|
||||
/* -- timer -- */
|
||||
unsigned int timer_resolution; /* timer resolution */
|
||||
|
||||
@@ -2209,9 +2205,8 @@ struct _snd_pcm_runtime {
|
||||
For the operators (callbacks) of each sound driver, most of
|
||||
these records are supposed to be read-only. Only the PCM
|
||||
middle-layer changes / updates them. The exceptions are
|
||||
the hardware description (hw), interrupt callbacks
|
||||
(transfer_ack_xxx), DMA buffer information, and the private
|
||||
data. Besides, if you use the standard buffer allocation
|
||||
the hardware description (hw) DMA buffer information and the
|
||||
private data. Besides, if you use the standard buffer allocation
|
||||
method via <function>snd_pcm_lib_malloc_pages()</function>,
|
||||
you don't need to set the DMA buffer information by yourself.
|
||||
</para>
|
||||
@@ -2538,16 +2533,6 @@ struct _snd_pcm_runtime {
|
||||
</para>
|
||||
</section>
|
||||
|
||||
<section id="pcm-interface-runtime-intr">
|
||||
<title>Interrupt Callbacks</title>
|
||||
<para>
|
||||
The field <structfield>transfer_ack_begin</structfield> and
|
||||
<structfield>transfer_ack_end</structfield> are called at
|
||||
the beginning and at the end of
|
||||
<function>snd_pcm_period_elapsed()</function>, respectively.
|
||||
</para>
|
||||
</section>
|
||||
|
||||
</section>
|
||||
|
||||
<section id="pcm-interface-operators">
|
||||
|
||||
@@ -0,0 +1,17 @@
|
||||
AK4613 I2C transmitter
|
||||
|
||||
This device supports I2C mode only.
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : "asahi-kasei,ak4613"
|
||||
- reg : The chip select number on the I2C bus
|
||||
|
||||
Example:
|
||||
|
||||
&i2c {
|
||||
ak4613: ak4613@0x10 {
|
||||
compatible = "asahi-kasei,ak4613";
|
||||
reg = <0x10>;
|
||||
};
|
||||
};
|
||||
@@ -7,7 +7,14 @@ Required properties:
|
||||
- compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
|
||||
- reg : The chip select number on the I2C bus
|
||||
|
||||
Example:
|
||||
Optional properties:
|
||||
|
||||
- #clock-cells : common clock binding; shall be set to 0
|
||||
- clocks : common clock binding; MCKI clock
|
||||
- clock-frequency : common clock binding; frequency of MCKO
|
||||
- clock-output-names : common clock binding; MCKO clock name
|
||||
|
||||
Example 1:
|
||||
|
||||
&i2c {
|
||||
ak4648: ak4648@0x12 {
|
||||
@@ -15,3 +22,16 @@ Example:
|
||||
reg = <0x12>;
|
||||
};
|
||||
};
|
||||
|
||||
Example 2:
|
||||
|
||||
&i2c {
|
||||
ak4643: codec@12 {
|
||||
compatible = "asahi-kasei,ak4643";
|
||||
reg = <0x12>;
|
||||
#clock-cells = <0>;
|
||||
clocks = <&audio_clock>;
|
||||
clock-frequency = <12288000>;
|
||||
clock-output-names = "ak4643_mcko";
|
||||
};
|
||||
};
|
||||
|
||||
@@ -0,0 +1,52 @@
|
||||
* Atmel ClassD driver under ALSA SoC architecture
|
||||
|
||||
Required properties:
|
||||
- compatible
|
||||
Should be "atmel,sama5d2-classd".
|
||||
- reg
|
||||
Should contain ClassD registers location and length.
|
||||
- interrupts
|
||||
Should contain the IRQ line for the ClassD.
|
||||
- dmas
|
||||
One DMA specifiers as described in atmel-dma.txt and dma.txt files.
|
||||
- dma-names
|
||||
Must be "tx".
|
||||
- clock-names
|
||||
Tuple listing input clock names.
|
||||
Required elements: "pclk", "gclk" and "aclk".
|
||||
- clocks
|
||||
Please refer to clock-bindings.txt.
|
||||
|
||||
Optional properties:
|
||||
- pinctrl-names, pinctrl-0
|
||||
Please refer to pinctrl-bindings.txt.
|
||||
- atmel,model
|
||||
The user-visible name of this sound complex.
|
||||
The default value is "CLASSD".
|
||||
- atmel,pwm-type
|
||||
PWM modulation type, "single" or "diff".
|
||||
The default value is "single".
|
||||
- atmel,non-overlap-time
|
||||
Set non-overlapping time, the unit is nanosecond(ns).
|
||||
There are four values,
|
||||
<5>, <10>, <15>, <20>, the default value is <10>.
|
||||
Non-overlapping will be disabled if not specified.
|
||||
|
||||
Example:
|
||||
classd: classd@fc048000 {
|
||||
compatible = "atmel,sama5d2-classd";
|
||||
reg = <0xfc048000 0x100>;
|
||||
interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>;
|
||||
dmas = <&dma0
|
||||
(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1)
|
||||
| AT91_XDMAC_DT_PERID(47))>;
|
||||
dma-names = "tx";
|
||||
clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>;
|
||||
clock-names = "pclk", "gclk", "aclk";
|
||||
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&pinctrl_classd_default>;
|
||||
atmel,model = "classd @ SAMA5D2-Xplained";
|
||||
atmel,pwm-type = "diff";
|
||||
atmel,non-overlap-time = <10>;
|
||||
};
|
||||
@@ -0,0 +1,41 @@
|
||||
Dialog Semiconductor DA7213 Audio Codec bindings
|
||||
|
||||
======
|
||||
|
||||
Required properties:
|
||||
- compatible : Should be "dlg,da7213"
|
||||
- reg: Specifies the I2C slave address
|
||||
|
||||
Optional properties:
|
||||
- clocks : phandle and clock specifier for codec MCLK.
|
||||
- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
|
||||
|
||||
- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1
|
||||
[<1600>, <2200>, <2500>, <3000>]
|
||||
- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2
|
||||
[<1600>, <2200>, <2500>, <3000>]
|
||||
- dlg,dmic-data-sel : DMIC channel select based on clock edge.
|
||||
["lrise_rfall", "lfall_rrise"]
|
||||
- dlg,dmic-samplephase : When to sample audio from DMIC.
|
||||
["on_clkedge", "between_clkedge"]
|
||||
- dlg,dmic-clkrate : DMIC clock frequency (Hz).
|
||||
[<1500000>, <3000000>]
|
||||
|
||||
======
|
||||
|
||||
Example:
|
||||
|
||||
codec_i2c: da7213@1a {
|
||||
compatible = "dlg,da7213";
|
||||
reg = <0x1a>;
|
||||
|
||||
clocks = <&clks 201>;
|
||||
clock-names = "mclk";
|
||||
|
||||
dlg,micbias1-lvl = <2500>;
|
||||
dlg,micbias2-lvl = <2500>;
|
||||
|
||||
dlg,dmic-data-sel = "lrise_rfall";
|
||||
dlg,dmic-samplephase = "between_clkedge";
|
||||
dlg,dmic-clkrate = <3000000>;
|
||||
};
|
||||
@@ -0,0 +1,106 @@
|
||||
Dialog Semiconductor DA7219 Audio Codec bindings
|
||||
|
||||
DA7219 is an audio codec with advanced accessory detect features.
|
||||
|
||||
======
|
||||
|
||||
Required properties:
|
||||
- compatible : Should be "dlg,da7219"
|
||||
- reg: Specifies the I2C slave address
|
||||
|
||||
- interrupt-parent : Specifies the phandle of the interrupt controller to which
|
||||
the IRQs from DA7219 are delivered to.
|
||||
- interrupts : IRQ line info for DA7219.
|
||||
(See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for
|
||||
further information relating to interrupt properties)
|
||||
|
||||
- VDD-supply: VDD power supply for the device
|
||||
- VDDMIC-supply: VDDMIC power supply for the device
|
||||
- VDDIO-supply: VDDIO power supply for the device
|
||||
(See Documentation/devicetree/bindings/regulator/regulator.txt for further
|
||||
information relating to regulators)
|
||||
|
||||
Optional properties:
|
||||
- interrupt-names : Name associated with interrupt line. Should be "wakeup" if
|
||||
interrupt is to be used to wake system, otherwise "irq" should be used.
|
||||
- wakeup-source: Flag to indicate this device can wake system (suspend/resume).
|
||||
|
||||
- clocks : phandle and clock specifier for codec MCLK.
|
||||
- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
|
||||
|
||||
- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine
|
||||
[<1050>, <1100>, <1200>, <1400>]
|
||||
- dlg,micbias-lvl : Voltage (mV) for Mic Bias
|
||||
[<1800>, <2000>, <2200>, <2400>, <2600>]
|
||||
- dlg,mic-amp-in-sel : Mic input source type
|
||||
["diff", "se_p", "se_n"]
|
||||
|
||||
======
|
||||
|
||||
Child node - 'da7219_aad':
|
||||
|
||||
Optional properties:
|
||||
- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV).
|
||||
[<2800>, <2900>]
|
||||
- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms)
|
||||
- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms)
|
||||
[<2>, <5>, <10>, <50>, <100>, <200>, <500>]
|
||||
- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms)
|
||||
[<200>, <500>, <750>, <1000>]
|
||||
- dlg,jack-ins-deb : Debounce time for jack insertion (ms)
|
||||
[<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>]
|
||||
- dlg,jack-det-rate: Jack type detection latency (3/4 pole)
|
||||
["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"]
|
||||
- dlg,jack-rem-deb : Debounce time for jack removal (ms)
|
||||
[<1>, <5>, <10>, <20>]
|
||||
- dlg,a-d-btn-thr : Impedance threshold between buttons A and D
|
||||
[0x0 - 0xFF]
|
||||
- dlg,d-b-btn-thr : Impedance threshold between buttons D and B
|
||||
[0x0 - 0xFF]
|
||||
- dlg,b-c-btn-thr : Impedance threshold between buttons B and C
|
||||
[0x0 - 0xFF]
|
||||
- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic
|
||||
[0x0 - 0xFF]
|
||||
- dlg,btn-avg : Number of 8-bit readings for averaged button measurement
|
||||
[<1>, <2>, <4>, <8>]
|
||||
- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement
|
||||
[<1>, <2>, <4>, <8>]
|
||||
|
||||
======
|
||||
|
||||
Example:
|
||||
|
||||
codec: da7219@1a {
|
||||
compatible = "dlg,da7219";
|
||||
reg = <0x1a>;
|
||||
|
||||
interrupt-parent = <&gpio6>;
|
||||
interrupts = <11 IRQ_TYPE_LEVEL_HIGH>;
|
||||
|
||||
VDD-supply = <®_audio>;
|
||||
VDDMIC-supply = <®_audio>;
|
||||
VDDIO-supply = <®_audio>;
|
||||
|
||||
clocks = <&clks 201>;
|
||||
clock-names = "mclk";
|
||||
|
||||
dlg,ldo-lvl = <1200>;
|
||||
dlg,micbias-lvl = <2600>;
|
||||
dlg,mic-amp-in-sel = "diff";
|
||||
|
||||
da7219_aad {
|
||||
dlg,btn-cfg = <50>;
|
||||
dlg,mic-det-thr = <500>;
|
||||
dlg,jack-ins-deb = <20>;
|
||||
dlg,jack-det-rate = "32ms_64ms";
|
||||
dlg,jack-rem-deb = <1>;
|
||||
|
||||
dlg,a-d-btn-thr = <0xa>;
|
||||
dlg,d-b-btn-thr = <0x16>;
|
||||
dlg,b-c-btn-thr = <0x21>;
|
||||
dlg,c-mic-btn-thr = <0x3E>;
|
||||
|
||||
dlg,btn-avg = <4>;
|
||||
dlg,adc-1bit-rpt = <1>;
|
||||
};
|
||||
};
|
||||
@@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit
|
||||
from the simplification of a new card support and the capability of the wide
|
||||
sample rates support through ASRC.
|
||||
|
||||
Note: The card is initially designed for those sound cards who use I2S and
|
||||
PCM DAI formats. However, it'll be also possible to support those non
|
||||
I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long
|
||||
as the driver has been properly upgraded.
|
||||
Note: The card is initially designed for those sound cards who use AC'97, I2S
|
||||
and PCM DAI formats. However, it'll be also possible to support those non
|
||||
AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
|
||||
long as the driver has been properly upgraded.
|
||||
|
||||
|
||||
The compatible list for this generic sound card currently:
|
||||
"fsl,imx-audio-ac97"
|
||||
|
||||
"fsl,imx-audio-cs42888"
|
||||
|
||||
"fsl,imx-audio-wm8962"
|
||||
|
||||
@@ -0,0 +1,102 @@
|
||||
Nuvoton NAU8825 audio codec
|
||||
|
||||
This device supports I2C only.
|
||||
|
||||
Required properties:
|
||||
- compatible : Must be "nuvoton,nau8825"
|
||||
|
||||
- reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1).
|
||||
|
||||
Optional properties:
|
||||
- nuvoton,jkdet-enable: Enable jack detection via JKDET pin.
|
||||
- nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled,
|
||||
otherwise pin in high impedance state.
|
||||
- nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down.
|
||||
- nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low.
|
||||
|
||||
- nuvoton,vref-impedance: VREF Impedance selection
|
||||
0 - Open
|
||||
1 - 25 kOhm
|
||||
2 - 125 kOhm
|
||||
3 - 2.5 kOhm
|
||||
|
||||
- nuvoton,micbias-voltage: Micbias voltage level.
|
||||
0 - VDDA
|
||||
1 - VDDA
|
||||
2 - VDDA * 1.1
|
||||
3 - VDDA * 1.2
|
||||
4 - VDDA * 1.3
|
||||
5 - VDDA * 1.4
|
||||
6 - VDDA * 1.53
|
||||
7 - VDDA * 1.53
|
||||
|
||||
- nuvoton,sar-threshold-num: Number of buttons supported
|
||||
- nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as
|
||||
SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R)
|
||||
where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance.
|
||||
Refer datasheet section 10.2 for more information about threshold calculation.
|
||||
|
||||
- nuvoton,sar-hysteresis: Button impedance measurement hysteresis.
|
||||
|
||||
- nuvoton,sar-voltage: Reference voltage for button impedance measurement.
|
||||
0 - VDDA
|
||||
1 - VDDA
|
||||
2 - VDDA * 1.1
|
||||
3 - VDDA * 1.2
|
||||
4 - VDDA * 1.3
|
||||
5 - VDDA * 1.4
|
||||
6 - VDDA * 1.53
|
||||
7 - VDDA * 1.53
|
||||
|
||||
- nuvoton,sar-compare-time: SAR compare time
|
||||
0 - 500 ns
|
||||
1 - 1 us
|
||||
2 - 2 us
|
||||
3 - 4 us
|
||||
|
||||
- nuvoton,sar-sampling-time: SAR sampling time
|
||||
0 - 2 us
|
||||
1 - 4 us
|
||||
2 - 8 us
|
||||
3 - 16 us
|
||||
|
||||
- nuvoton,short-key-debounce: Button short key press debounce time.
|
||||
0 - 30 ms
|
||||
1 - 50 ms
|
||||
2 - 100 ms
|
||||
3 - 30 ms
|
||||
|
||||
- nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
|
||||
- nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms
|
||||
|
||||
- clocks: list of phandle and clock specifier pairs according to common clock bindings for the
|
||||
clocks described in clock-names
|
||||
- clock-names: should include "mclk" for the MCLK master clock
|
||||
|
||||
Example:
|
||||
|
||||
headset: nau8825@1a {
|
||||
compatible = "nuvoton,nau8825";
|
||||
reg = <0x1a>;
|
||||
interrupt-parent = <&gpio>;
|
||||
interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>;
|
||||
nuvoton,jkdet-enable;
|
||||
nuvoton,jkdet-pull-enable;
|
||||
nuvoton,jkdet-pull-up;
|
||||
nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>;
|
||||
nuvoton,vref-impedance = <2>;
|
||||
nuvoton,micbias-voltage = <6>;
|
||||
// Setup 4 buttons impedance according to Android specification
|
||||
nuvoton,sar-threshold-num = <4>;
|
||||
nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
|
||||
nuvoton,sar-hysteresis = <1>;
|
||||
nuvoton,sar-voltage = <0>;
|
||||
nuvoton,sar-compare-time = <0>;
|
||||
nuvoton,sar-sampling-time = <0>;
|
||||
nuvoton,short-key-debounce = <2>;
|
||||
nuvoton,jack-insert-debounce = <7>;
|
||||
nuvoton,jack-eject-debounce = <7>;
|
||||
|
||||
clock-names = "mclk";
|
||||
clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;
|
||||
};
|
||||
@@ -4,10 +4,12 @@ Required properties:
|
||||
- compatible : "renesas,rcar_sound-<soctype>", fallbacks
|
||||
"renesas,rcar_sound-gen1" if generation1, and
|
||||
"renesas,rcar_sound-gen2" if generation2
|
||||
"renesas,rcar_sound-gen3" if generation3
|
||||
Examples with soctypes are:
|
||||
- "renesas,rcar_sound-r8a7778" (R-Car M1A)
|
||||
- "renesas,rcar_sound-r8a7790" (R-Car H2)
|
||||
- "renesas,rcar_sound-r8a7791" (R-Car M2-W)
|
||||
- "renesas,rcar_sound-r8a7795" (R-Car H3)
|
||||
- reg : Should contain the register physical address.
|
||||
required register is
|
||||
SRU/ADG/SSI if generation1
|
||||
@@ -30,6 +32,11 @@ Required properties:
|
||||
- rcar_sound,dai : DAI contents.
|
||||
The number of DAI subnode should be same as HW.
|
||||
see below for detail.
|
||||
- #sound-dai-cells : it must be 0 if your system is using single DAI
|
||||
it must be 1 if your system is using multi DAI
|
||||
- #clock-cells : it must be 0 if your system has audio_clkout
|
||||
it must be 1 if your system has audio_clkout0/1/2/3
|
||||
- clock-frequency : for all audio_clkout0/1/2/3
|
||||
|
||||
SSI subnode properties:
|
||||
- interrupts : Should contain SSI interrupt for PIO transfer
|
||||
|
||||
@@ -12,8 +12,6 @@ Required properties:
|
||||
- reg: physical base address of the controller and length of memory mapped
|
||||
region.
|
||||
- interrupts: should contain the I2S interrupt.
|
||||
- #address-cells: should be 1.
|
||||
- #size-cells: should be 0.
|
||||
- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
|
||||
Documentation/devicetree/bindings/dma/dma.txt
|
||||
- dma-names: should include "tx" and "rx".
|
||||
@@ -21,6 +19,7 @@ Required properties:
|
||||
- clock-names: should contain followings:
|
||||
- "i2s_hclk": clock for I2S BUS
|
||||
- "i2s_clk" : clock for I2S controller
|
||||
- rockchip,capture-channels: max capture channels, if not set, 2 channels default.
|
||||
|
||||
Example for rk3288 I2S controller:
|
||||
|
||||
@@ -28,10 +27,9 @@ i2s@ff890000 {
|
||||
compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
|
||||
reg = <0xff890000 0x10000>;
|
||||
interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
dmas = <&pdma1 0>, <&pdma1 1>;
|
||||
dma-names = "tx", "rx";
|
||||
clock-names = "i2s_hclk", "i2s_clk";
|
||||
clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
|
||||
rockchip,capture-channels = <2>;
|
||||
};
|
||||
|
||||
@@ -0,0 +1,40 @@
|
||||
* Rockchip SPDIF transceiver
|
||||
|
||||
The S/PDIF audio block is a stereo transceiver that allows the
|
||||
processor to receive and transmit digital audio via an coaxial cable or
|
||||
a fibre cable.
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible: should be one of the following:
|
||||
- "rockchip,rk3288-spdif", "rockchip,rk3188-spdif" or
|
||||
"rockchip,rk3066-spdif"
|
||||
- reg: physical base address of the controller and length of memory mapped
|
||||
region.
|
||||
- interrupts: should contain the SPDIF interrupt.
|
||||
- dmas: DMA specifiers for tx dma. See the DMA client binding,
|
||||
Documentation/devicetree/bindings/dma/dma.txt
|
||||
- dma-names: should be "tx"
|
||||
- clocks: a list of phandle + clock-specifier pairs, one for each entry
|
||||
in clock-names.
|
||||
- clock-names: should contain following:
|
||||
- "hclk": clock for SPDIF controller
|
||||
- "mclk" : clock for SPDIF bus
|
||||
|
||||
Required properties on RK3288:
|
||||
- rockchip,grf: the phandle of the syscon node for the general register
|
||||
file (GRF)
|
||||
|
||||
Example for the rk3188 SPDIF controller:
|
||||
|
||||
spdif: spdif@0x1011e000 {
|
||||
compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
|
||||
reg = <0x1011e000 0x2000>;
|
||||
interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
|
||||
dmas = <&dmac1_s 8>;
|
||||
dma-names = "tx";
|
||||
clock-names = "hclk", "mclk";
|
||||
clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>;
|
||||
status = "disabled";
|
||||
#sound-dai-cells = <0>;
|
||||
};
|
||||
@@ -14,7 +14,8 @@ Optional properties:
|
||||
|
||||
- realtek,in1-differential
|
||||
- realtek,in2-differential
|
||||
Boolean. Indicate MIC1/2 input are differential, rather than single-ended.
|
||||
- realtek,in3-differential
|
||||
Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended.
|
||||
|
||||
- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
|
||||
|
||||
@@ -24,9 +25,11 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640:
|
||||
* DMIC2
|
||||
* MICBIAS1
|
||||
* IN1P
|
||||
* IN1R
|
||||
* IN1N
|
||||
* IN2P
|
||||
* IN2R
|
||||
* IN2N
|
||||
* IN3P
|
||||
* IN3N
|
||||
* HPOL
|
||||
* HPOR
|
||||
* LOUTL
|
||||
|
||||
@@ -0,0 +1,27 @@
|
||||
* Allwinner A10 Codec
|
||||
|
||||
Required properties:
|
||||
- compatible: must be either "allwinner,sun4i-a10-codec" or
|
||||
"allwinner,sun7i-a20-codec"
|
||||
- reg: must contain the registers location and length
|
||||
- interrupts: must contain the codec interrupt
|
||||
- dmas: DMA channels for tx and rx dma. See the DMA client binding,
|
||||
Documentation/devicetree/bindings/dma/dma.txt
|
||||
- dma-names: should include "tx" and "rx".
|
||||
- clocks: a list of phandle + clock-specifer pairs, one for each entry
|
||||
in clock-names.
|
||||
- clock-names: should contain followings:
|
||||
- "apb": the parent APB clock for this controller
|
||||
- "codec": the parent module clock
|
||||
|
||||
Example:
|
||||
codec: codec@01c22c00 {
|
||||
#sound-dai-cells = <0>;
|
||||
compatible = "allwinner,sun7i-a20-codec";
|
||||
reg = <0x01c22c00 0x40>;
|
||||
interrupts = <0 30 4>;
|
||||
clocks = <&apb0_gates 0>, <&codec_clk>;
|
||||
clock-names = "apb", "codec";
|
||||
dmas = <&dma 0 19>, <&dma 0 19>;
|
||||
dma-names = "rx", "tx";
|
||||
};
|
||||
@@ -4,11 +4,15 @@ This specifies audio DAI's TDM slot.
|
||||
|
||||
TDM slot properties:
|
||||
dai-tdm-slot-num : Number of slots in use.
|
||||
dai-tdm-slot-width : Width in bits for each slot.
|
||||
dai-tdm-slot-width : Width in bits for each slot.
|
||||
dai-tdm-slot-tx-mask : Transmit direction slot mask, optional
|
||||
dai-tdm-slot-rx-mask : Receive direction slot mask, optional
|
||||
|
||||
For instance:
|
||||
dai-tdm-slot-num = <2>;
|
||||
dai-tdm-slot-width = <8>;
|
||||
dai-tdm-slot-tx-mask = <0 1>;
|
||||
dai-tdm-slot-rx-mask = <1 0>;
|
||||
|
||||
And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
|
||||
to specify a explicit mapping of the channels and the slots. If it's absent
|
||||
@@ -18,3 +22,8 @@ tx and rx masks.
|
||||
For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
|
||||
for an active slot as default, and the default active bits are at the LSB of
|
||||
the masks.
|
||||
|
||||
The explicit masks are given as array of integers, where the first
|
||||
number presents bit-0 (LSB), second presents bit-1, etc. Any non zero
|
||||
number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask()
|
||||
does not do anything, if either mask is set non zero value.
|
||||
|
||||
@@ -1,322 +0,0 @@
|
||||
Notes on Universal Interface for Intel High Definition Audio Codec
|
||||
------------------------------------------------------------------
|
||||
|
||||
Takashi Iwai <tiwai@suse.de>
|
||||
|
||||
|
||||
[Still a draft version]
|
||||
|
||||
|
||||
General
|
||||
=======
|
||||
|
||||
The snd-hda-codec module supports the generic access function for the
|
||||
High Definition (HD) audio codecs. It's designed to be independent
|
||||
from the controller code like ac97 codec module. The real accessors
|
||||
from/to the controller must be implemented in the lowlevel driver.
|
||||
|
||||
The structure of this module is similar with ac97_codec module.
|
||||
Each codec chip belongs to a bus class which communicates with the
|
||||
controller.
|
||||
|
||||
|
||||
Initialization of Bus Instance
|
||||
==============================
|
||||
|
||||
The card driver has to create struct hda_bus at first. The template
|
||||
struct should be filled and passed to the constructor:
|
||||
|
||||
struct hda_bus_template {
|
||||
void *private_data;
|
||||
struct pci_dev *pci;
|
||||
const char *modelname;
|
||||
struct hda_bus_ops ops;
|
||||
};
|
||||
|
||||
The card driver can set and use the private_data field to retrieve its
|
||||
own data in callback functions. The pci field is used when the patch
|
||||
needs to check the PCI subsystem IDs, so on. For non-PCI system, it
|
||||
doesn't have to be set, of course.
|
||||
The modelname field specifies the board's specific configuration. The
|
||||
string is passed to the codec parser, and it depends on the parser how
|
||||
the string is used.
|
||||
These fields, private_data, pci and modelname are all optional.
|
||||
|
||||
The ops field contains the callback functions as the following:
|
||||
|
||||
struct hda_bus_ops {
|
||||
int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
|
||||
unsigned int verb, unsigned int parm);
|
||||
unsigned int (*get_response)(struct hda_codec *codec);
|
||||
void (*private_free)(struct hda_bus *);
|
||||
#ifdef CONFIG_SND_HDA_POWER_SAVE
|
||||
void (*pm_notify)(struct hda_codec *codec);
|
||||
#endif
|
||||
};
|
||||
|
||||
The command callback is called when the codec module needs to send a
|
||||
VERB to the controller. It's always a single command.
|
||||
The get_response callback is called when the codec requires the answer
|
||||
for the last command. These two callbacks are mandatory and have to
|
||||
be given.
|
||||
The third, private_free callback, is optional. It's called in the
|
||||
destructor to release any necessary data in the lowlevel driver.
|
||||
|
||||
The pm_notify callback is available only with
|
||||
CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs
|
||||
to power up or may power down. The controller should check the all
|
||||
belonging codecs on the bus whether they are actually powered off
|
||||
(check codec->power_on), and optionally the driver may power down the
|
||||
controller side, too.
|
||||
|
||||
The bus instance is created via snd_hda_bus_new(). You need to pass
|
||||
the card instance, the template, and the pointer to store the
|
||||
resultant bus instance.
|
||||
|
||||
int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
|
||||
struct hda_bus **busp);
|
||||
|
||||
It returns zero if successful. A negative return value means any
|
||||
error during creation.
|
||||
|
||||
|
||||
Creation of Codec Instance
|
||||
==========================
|
||||
|
||||
Each codec chip on the board is then created on the BUS instance.
|
||||
To create a codec instance, call snd_hda_codec_new().
|
||||
|
||||
int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
|
||||
struct hda_codec **codecp);
|
||||
|
||||
The first argument is the BUS instance, the second argument is the
|
||||
address of the codec, and the last one is the pointer to store the
|
||||
resultant codec instance (can be NULL if not needed).
|
||||
|
||||
The codec is stored in a linked list of bus instance. You can follow
|
||||
the codec list like:
|
||||
|
||||
struct hda_codec *codec;
|
||||
list_for_each_entry(codec, &bus->codec_list, list) {
|
||||
...
|
||||
}
|
||||
|
||||
The codec isn't initialized at this stage properly. The
|
||||
initialization sequence is called when the controls are built later.
|
||||
|
||||
|
||||
Codec Access
|
||||
============
|
||||
|
||||
To access codec, use snd_hda_codec_read() and snd_hda_codec_write().
|
||||
snd_hda_param_read() is for reading parameters.
|
||||
For writing a sequence of verbs, use snd_hda_sequence_write().
|
||||
|
||||
There are variants of cached read/write, snd_hda_codec_write_cache(),
|
||||
snd_hda_sequence_write_cache(). These are used for recording the
|
||||
register states for the power-management resume. When no PM is needed,
|
||||
these are equivalent with non-cached version.
|
||||
|
||||
To retrieve the number of sub nodes connected to the given node, use
|
||||
snd_hda_get_sub_nodes(). The connection list can be obtained via
|
||||
snd_hda_get_connections() call.
|
||||
|
||||
When an unsolicited event happens, pass the event via
|
||||
snd_hda_queue_unsol_event() so that the codec routines will process it
|
||||
later.
|
||||
|
||||
|
||||
(Mixer) Controls
|
||||
================
|
||||
|
||||
To create mixer controls of all codecs, call
|
||||
snd_hda_build_controls(). It then builds the mixers and does
|
||||
initialization stuff on each codec.
|
||||
|
||||
|
||||
PCM Stuff
|
||||
=========
|
||||
|
||||
snd_hda_build_pcms() gives the necessary information to create PCM
|
||||
streams. When it's called, each codec belonging to the bus stores
|
||||
codec->num_pcms and codec->pcm_info fields. The num_pcms indicates
|
||||
the number of elements in pcm_info array. The card driver is supposed
|
||||
to traverse the codec linked list, read the pcm information in
|
||||
pcm_info array, and build pcm instances according to them.
|
||||
|
||||
The pcm_info array contains the following record:
|
||||
|
||||
/* PCM information for each substream */
|
||||
struct hda_pcm_stream {
|
||||
unsigned int substreams; /* number of substreams, 0 = not exist */
|
||||
unsigned int channels_min; /* min. number of channels */
|
||||
unsigned int channels_max; /* max. number of channels */
|
||||
hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */
|
||||
u32 rates; /* supported rates */
|
||||
u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */
|
||||
unsigned int maxbps; /* supported max. bit per sample */
|
||||
struct hda_pcm_ops ops;
|
||||
};
|
||||
|
||||
/* for PCM creation */
|
||||
struct hda_pcm {
|
||||
char *name;
|
||||
struct hda_pcm_stream stream[2];
|
||||
};
|
||||
|
||||
The name can be passed to snd_pcm_new(). The stream field contains
|
||||
the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and
|
||||
capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver
|
||||
should pass substreams to snd_pcm_new() for the number of substreams
|
||||
to create.
|
||||
|
||||
The channels_min, channels_max, rates and formats should be copied to
|
||||
runtime->hw record. They and maxbps fields are used also to compute
|
||||
the format value for the HDA codec and controller. Call
|
||||
snd_hda_calc_stream_format() to get the format value.
|
||||
|
||||
The ops field contains the following callback functions:
|
||||
|
||||
struct hda_pcm_ops {
|
||||
int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
|
||||
struct snd_pcm_substream *substream);
|
||||
int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
|
||||
struct snd_pcm_substream *substream);
|
||||
int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
|
||||
unsigned int stream_tag, unsigned int format,
|
||||
struct snd_pcm_substream *substream);
|
||||
int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
|
||||
struct snd_pcm_substream *substream);
|
||||
};
|
||||
|
||||
All are non-NULL, so you can call them safely without NULL check.
|
||||
|
||||
The open callback should be called in PCM open after runtime->hw is
|
||||
set up. It may override some setting and constraints additionally.
|
||||
Similarly, the close callback should be called in the PCM close.
|
||||
|
||||
The prepare callback should be called in PCM prepare. This will set
|
||||
up the codec chip properly for the operation. The cleanup should be
|
||||
called in hw_free to clean up the configuration.
|
||||
|
||||
The caller should check the return value, at least for open and
|
||||
prepare callbacks. When a negative value is returned, some error
|
||||
occurred.
|
||||
|
||||
|
||||
Proc Files
|
||||
==========
|
||||
|
||||
Each codec dumps the widget node information in
|
||||
/proc/asound/card*/codec#* file. This information would be really
|
||||
helpful for debugging. Please provide its contents together with the
|
||||
bug report.
|
||||
|
||||
|
||||
Power Management
|
||||
================
|
||||
|
||||
It's simple:
|
||||
Call snd_hda_suspend() in the PM suspend callback.
|
||||
Call snd_hda_resume() in the PM resume callback.
|
||||
|
||||
|
||||
Codec Preset (Patch)
|
||||
====================
|
||||
|
||||
To set up and handle the codec functionality fully, each codec may
|
||||
have a codec preset (patch). It's defined in struct hda_codec_preset:
|
||||
|
||||
struct hda_codec_preset {
|
||||
unsigned int id;
|
||||
unsigned int mask;
|
||||
unsigned int subs;
|
||||
unsigned int subs_mask;
|
||||
unsigned int rev;
|
||||
const char *name;
|
||||
int (*patch)(struct hda_codec *codec);
|
||||
};
|
||||
|
||||
When the codec id and codec subsystem id match with the given id and
|
||||
subs fields bitwise (with bitmask mask and subs_mask), the callback
|
||||
patch is called. The patch callback should initialize the codec and
|
||||
set the codec->patch_ops field. This is defined as below:
|
||||
|
||||
struct hda_codec_ops {
|
||||
int (*build_controls)(struct hda_codec *codec);
|
||||
int (*build_pcms)(struct hda_codec *codec);
|
||||
int (*init)(struct hda_codec *codec);
|
||||
void (*free)(struct hda_codec *codec);
|
||||
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
|
||||
#ifdef CONFIG_PM
|
||||
int (*suspend)(struct hda_codec *codec, pm_message_t state);
|
||||
int (*resume)(struct hda_codec *codec);
|
||||
#endif
|
||||
#ifdef CONFIG_SND_HDA_POWER_SAVE
|
||||
int (*check_power_status)(struct hda_codec *codec,
|
||||
hda_nid_t nid);
|
||||
#endif
|
||||
};
|
||||
|
||||
The build_controls callback is called from snd_hda_build_controls().
|
||||
Similarly, the build_pcms callback is called from
|
||||
snd_hda_build_pcms(). The init callback is called after
|
||||
build_controls to initialize the hardware.
|
||||
The free callback is called as a destructor.
|
||||
|
||||
The unsol_event callback is called when an unsolicited event is
|
||||
received.
|
||||
|
||||
The suspend and resume callbacks are for power management.
|
||||
They can be NULL if no special sequence is required. When the resume
|
||||
callback is NULL, the driver calls the init callback and resumes the
|
||||
registers from the cache. If other handling is needed, you'd need to
|
||||
write your own resume callback. There, the amp values can be resumed
|
||||
via
|
||||
void snd_hda_codec_resume_amp(struct hda_codec *codec);
|
||||
and the other codec registers via
|
||||
void snd_hda_codec_resume_cache(struct hda_codec *codec);
|
||||
|
||||
The check_power_status callback is called when the amp value of the
|
||||
given widget NID is changed. The codec code can turn on/off the power
|
||||
appropriately from this information.
|
||||
|
||||
Each entry can be NULL if not necessary to be called.
|
||||
|
||||
|
||||
Generic Parser
|
||||
==============
|
||||
|
||||
When the device doesn't match with any given presets, the widgets are
|
||||
parsed via th generic parser (hda_generic.c). Its support is
|
||||
limited: no multi-channel support, for example.
|
||||
|
||||
|
||||
Digital I/O
|
||||
===========
|
||||
|
||||
Call snd_hda_create_spdif_out_ctls() from the patch to create controls
|
||||
related with SPDIF out.
|
||||
|
||||
|
||||
Helper Functions
|
||||
================
|
||||
|
||||
snd_hda_get_codec_name() stores the codec name on the given string.
|
||||
|
||||
snd_hda_check_board_config() can be used to obtain the configuration
|
||||
information matching with the device. Define the model string table
|
||||
and the table with struct snd_pci_quirk entries (zero-terminated),
|
||||
and pass it to the function. The function checks the modelname given
|
||||
as a module parameter, and PCI subsystem IDs. If the matching entry
|
||||
is found, it returns the config field value.
|
||||
|
||||
snd_hda_add_new_ctls() can be used to create and add control entries.
|
||||
Pass the zero-terminated array of struct snd_kcontrol_new
|
||||
|
||||
Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be
|
||||
used for the entry of struct snd_kcontrol_new.
|
||||
|
||||
The input MUX helper callbacks for such a control are provided, too:
|
||||
snd_hda_input_mux_info() and snd_hda_input_mux_put(). See
|
||||
patch_realtek.c for example.
|
||||
@@ -3368,6 +3368,7 @@ M: Support Opensource <support.opensource@diasemi.com>
|
||||
W: http://www.dialog-semiconductor.com/products
|
||||
S: Supported
|
||||
F: Documentation/hwmon/da90??
|
||||
F: Documentation/devicetree/bindings/sound/da[79]*.txt
|
||||
F: drivers/gpio/gpio-da90??.c
|
||||
F: drivers/hwmon/da90??-hwmon.c
|
||||
F: drivers/iio/adc/da91??-*.c
|
||||
|
||||
@@ -832,6 +832,7 @@ int i915_driver_load(struct drm_device *dev, unsigned long flags)
|
||||
mutex_init(&dev_priv->sb_lock);
|
||||
mutex_init(&dev_priv->modeset_restore_lock);
|
||||
mutex_init(&dev_priv->csr_lock);
|
||||
mutex_init(&dev_priv->av_mutex);
|
||||
|
||||
intel_pm_setup(dev);
|
||||
|
||||
|
||||
@@ -1885,6 +1885,11 @@ struct drm_i915_private {
|
||||
/* hda/i915 audio component */
|
||||
struct i915_audio_component *audio_component;
|
||||
bool audio_component_registered;
|
||||
/**
|
||||
* av_mutex - mutex for audio/video sync
|
||||
*
|
||||
*/
|
||||
struct mutex av_mutex;
|
||||
|
||||
uint32_t hw_context_size;
|
||||
struct list_head context_list;
|
||||
|
||||
@@ -68,6 +68,31 @@ static const struct {
|
||||
{ 148500, AUD_CONFIG_PIXEL_CLOCK_HDMI_148500 },
|
||||
};
|
||||
|
||||
/* HDMI N/CTS table */
|
||||
#define TMDS_297M 297000
|
||||
#define TMDS_296M DIV_ROUND_UP(297000 * 1000, 1001)
|
||||
static const struct {
|
||||
int sample_rate;
|
||||
int clock;
|
||||
int n;
|
||||
int cts;
|
||||
} aud_ncts[] = {
|
||||
{ 44100, TMDS_296M, 4459, 234375 },
|
||||
{ 44100, TMDS_297M, 4704, 247500 },
|
||||
{ 48000, TMDS_296M, 5824, 281250 },
|
||||
{ 48000, TMDS_297M, 5120, 247500 },
|
||||
{ 32000, TMDS_296M, 5824, 421875 },
|
||||
{ 32000, TMDS_297M, 3072, 222750 },
|
||||
{ 88200, TMDS_296M, 8918, 234375 },
|
||||
{ 88200, TMDS_297M, 9408, 247500 },
|
||||
{ 96000, TMDS_296M, 11648, 281250 },
|
||||
{ 96000, TMDS_297M, 10240, 247500 },
|
||||
{ 176400, TMDS_296M, 17836, 234375 },
|
||||
{ 176400, TMDS_297M, 18816, 247500 },
|
||||
{ 192000, TMDS_296M, 23296, 281250 },
|
||||
{ 192000, TMDS_297M, 20480, 247500 },
|
||||
};
|
||||
|
||||
/* get AUD_CONFIG_PIXEL_CLOCK_HDMI_* value for mode */
|
||||
static u32 audio_config_hdmi_pixel_clock(struct drm_display_mode *mode)
|
||||
{
|
||||
@@ -90,6 +115,45 @@ static u32 audio_config_hdmi_pixel_clock(struct drm_display_mode *mode)
|
||||
return hdmi_audio_clock[i].config;
|
||||
}
|
||||
|
||||
static int audio_config_get_n(const struct drm_display_mode *mode, int rate)
|
||||
{
|
||||
int i;
|
||||
|
||||
for (i = 0; i < ARRAY_SIZE(aud_ncts); i++) {
|
||||
if ((rate == aud_ncts[i].sample_rate) &&
|
||||
(mode->clock == aud_ncts[i].clock)) {
|
||||
return aud_ncts[i].n;
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
static uint32_t audio_config_setup_n_reg(int n, uint32_t val)
|
||||
{
|
||||
int n_low, n_up;
|
||||
uint32_t tmp = val;
|
||||
|
||||
n_low = n & 0xfff;
|
||||
n_up = (n >> 12) & 0xff;
|
||||
tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK);
|
||||
tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) |
|
||||
(n_low << AUD_CONFIG_LOWER_N_SHIFT) |
|
||||
AUD_CONFIG_N_PROG_ENABLE);
|
||||
return tmp;
|
||||
}
|
||||
|
||||
/* check whether N/CTS/M need be set manually */
|
||||
static bool audio_rate_need_prog(struct intel_crtc *crtc,
|
||||
const struct drm_display_mode *mode)
|
||||
{
|
||||
if (((mode->clock == TMDS_297M) ||
|
||||
(mode->clock == TMDS_296M)) &&
|
||||
intel_pipe_has_type(crtc, INTEL_OUTPUT_HDMI))
|
||||
return true;
|
||||
else
|
||||
return false;
|
||||
}
|
||||
|
||||
static bool intel_eld_uptodate(struct drm_connector *connector,
|
||||
int reg_eldv, uint32_t bits_eldv,
|
||||
int reg_elda, uint32_t bits_elda,
|
||||
@@ -184,6 +248,8 @@ static void hsw_audio_codec_disable(struct intel_encoder *encoder)
|
||||
|
||||
DRM_DEBUG_KMS("Disable audio codec on pipe %c\n", pipe_name(pipe));
|
||||
|
||||
mutex_lock(&dev_priv->av_mutex);
|
||||
|
||||
/* Disable timestamps */
|
||||
tmp = I915_READ(HSW_AUD_CFG(pipe));
|
||||
tmp &= ~AUD_CONFIG_N_VALUE_INDEX;
|
||||
@@ -199,6 +265,8 @@ static void hsw_audio_codec_disable(struct intel_encoder *encoder)
|
||||
tmp &= ~AUDIO_ELD_VALID(pipe);
|
||||
tmp &= ~AUDIO_OUTPUT_ENABLE(pipe);
|
||||
I915_WRITE(HSW_AUD_PIN_ELD_CP_VLD, tmp);
|
||||
|
||||
mutex_unlock(&dev_priv->av_mutex);
|
||||
}
|
||||
|
||||
static void hsw_audio_codec_enable(struct drm_connector *connector,
|
||||
@@ -208,13 +276,20 @@ static void hsw_audio_codec_enable(struct drm_connector *connector,
|
||||
struct drm_i915_private *dev_priv = connector->dev->dev_private;
|
||||
struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc);
|
||||
enum pipe pipe = intel_crtc->pipe;
|
||||
struct i915_audio_component *acomp = dev_priv->audio_component;
|
||||
const uint8_t *eld = connector->eld;
|
||||
struct intel_digital_port *intel_dig_port =
|
||||
enc_to_dig_port(&encoder->base);
|
||||
enum port port = intel_dig_port->port;
|
||||
uint32_t tmp;
|
||||
int len, i;
|
||||
int n, rate;
|
||||
|
||||
DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n",
|
||||
pipe_name(pipe), drm_eld_size(eld));
|
||||
|
||||
mutex_lock(&dev_priv->av_mutex);
|
||||
|
||||
/* Enable audio presence detect, invalidate ELD */
|
||||
tmp = I915_READ(HSW_AUD_PIN_ELD_CP_VLD);
|
||||
tmp |= AUDIO_OUTPUT_ENABLE(pipe);
|
||||
@@ -246,13 +321,32 @@ static void hsw_audio_codec_enable(struct drm_connector *connector,
|
||||
/* Enable timestamps */
|
||||
tmp = I915_READ(HSW_AUD_CFG(pipe));
|
||||
tmp &= ~AUD_CONFIG_N_VALUE_INDEX;
|
||||
tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
|
||||
tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK;
|
||||
if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT))
|
||||
tmp |= AUD_CONFIG_N_VALUE_INDEX;
|
||||
else
|
||||
tmp |= audio_config_hdmi_pixel_clock(mode);
|
||||
|
||||
tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
|
||||
if (audio_rate_need_prog(intel_crtc, mode)) {
|
||||
if (!acomp)
|
||||
rate = 0;
|
||||
else if (port >= PORT_A && port <= PORT_E)
|
||||
rate = acomp->aud_sample_rate[port];
|
||||
else {
|
||||
DRM_ERROR("invalid port: %d\n", port);
|
||||
rate = 0;
|
||||
}
|
||||
n = audio_config_get_n(mode, rate);
|
||||
if (n != 0)
|
||||
tmp = audio_config_setup_n_reg(n, tmp);
|
||||
else
|
||||
DRM_DEBUG_KMS("no suitable N value is found\n");
|
||||
}
|
||||
|
||||
I915_WRITE(HSW_AUD_CFG(pipe), tmp);
|
||||
|
||||
mutex_unlock(&dev_priv->av_mutex);
|
||||
}
|
||||
|
||||
static void ilk_audio_codec_disable(struct intel_encoder *encoder)
|
||||
@@ -527,12 +621,91 @@ static int i915_audio_component_get_cdclk_freq(struct device *dev)
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int i915_audio_component_sync_audio_rate(struct device *dev,
|
||||
int port, int rate)
|
||||
{
|
||||
struct drm_i915_private *dev_priv = dev_to_i915(dev);
|
||||
struct drm_device *drm_dev = dev_priv->dev;
|
||||
struct intel_encoder *intel_encoder;
|
||||
struct intel_digital_port *intel_dig_port;
|
||||
struct intel_crtc *crtc;
|
||||
struct drm_display_mode *mode;
|
||||
struct i915_audio_component *acomp = dev_priv->audio_component;
|
||||
enum pipe pipe = -1;
|
||||
u32 tmp;
|
||||
int n;
|
||||
|
||||
/* HSW, BDW SKL need this fix */
|
||||
if (!IS_SKYLAKE(dev_priv) &&
|
||||
!IS_BROADWELL(dev_priv) &&
|
||||
!IS_HASWELL(dev_priv))
|
||||
return 0;
|
||||
|
||||
mutex_lock(&dev_priv->av_mutex);
|
||||
/* 1. get the pipe */
|
||||
for_each_intel_encoder(drm_dev, intel_encoder) {
|
||||
if (intel_encoder->type != INTEL_OUTPUT_HDMI)
|
||||
continue;
|
||||
intel_dig_port = enc_to_dig_port(&intel_encoder->base);
|
||||
if (port == intel_dig_port->port) {
|
||||
crtc = to_intel_crtc(intel_encoder->base.crtc);
|
||||
if (!crtc) {
|
||||
DRM_DEBUG_KMS("%s: crtc is NULL\n", __func__);
|
||||
continue;
|
||||
}
|
||||
pipe = crtc->pipe;
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
if (pipe == INVALID_PIPE) {
|
||||
DRM_DEBUG_KMS("no pipe for the port %c\n", port_name(port));
|
||||
mutex_unlock(&dev_priv->av_mutex);
|
||||
return -ENODEV;
|
||||
}
|
||||
DRM_DEBUG_KMS("pipe %c connects port %c\n",
|
||||
pipe_name(pipe), port_name(port));
|
||||
mode = &crtc->config->base.adjusted_mode;
|
||||
|
||||
/* port must be valid now, otherwise the pipe will be invalid */
|
||||
acomp->aud_sample_rate[port] = rate;
|
||||
|
||||
/* 2. check whether to set the N/CTS/M manually or not */
|
||||
if (!audio_rate_need_prog(crtc, mode)) {
|
||||
tmp = I915_READ(HSW_AUD_CFG(pipe));
|
||||
tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
|
||||
I915_WRITE(HSW_AUD_CFG(pipe), tmp);
|
||||
mutex_unlock(&dev_priv->av_mutex);
|
||||
return 0;
|
||||
}
|
||||
|
||||
n = audio_config_get_n(mode, rate);
|
||||
if (n == 0) {
|
||||
DRM_DEBUG_KMS("Using automatic mode for N value on port %c\n",
|
||||
port_name(port));
|
||||
tmp = I915_READ(HSW_AUD_CFG(pipe));
|
||||
tmp &= ~AUD_CONFIG_N_PROG_ENABLE;
|
||||
I915_WRITE(HSW_AUD_CFG(pipe), tmp);
|
||||
mutex_unlock(&dev_priv->av_mutex);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* 3. set the N/CTS/M */
|
||||
tmp = I915_READ(HSW_AUD_CFG(pipe));
|
||||
tmp = audio_config_setup_n_reg(n, tmp);
|
||||
I915_WRITE(HSW_AUD_CFG(pipe), tmp);
|
||||
|
||||
mutex_unlock(&dev_priv->av_mutex);
|
||||
return 0;
|
||||
}
|
||||
|
||||
static const struct i915_audio_component_ops i915_audio_component_ops = {
|
||||
.owner = THIS_MODULE,
|
||||
.get_power = i915_audio_component_get_power,
|
||||
.put_power = i915_audio_component_put_power,
|
||||
.codec_wake_override = i915_audio_component_codec_wake_override,
|
||||
.get_cdclk_freq = i915_audio_component_get_cdclk_freq,
|
||||
.sync_audio_rate = i915_audio_component_sync_audio_rate,
|
||||
};
|
||||
|
||||
static int i915_audio_component_bind(struct device *i915_dev,
|
||||
@@ -540,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev,
|
||||
{
|
||||
struct i915_audio_component *acomp = data;
|
||||
struct drm_i915_private *dev_priv = dev_to_i915(i915_dev);
|
||||
int i;
|
||||
|
||||
if (WARN_ON(acomp->ops || acomp->dev))
|
||||
return -EEXIST;
|
||||
@@ -547,6 +721,9 @@ static int i915_audio_component_bind(struct device *i915_dev,
|
||||
drm_modeset_lock_all(dev_priv->dev);
|
||||
acomp->ops = &i915_audio_component_ops;
|
||||
acomp->dev = i915_dev;
|
||||
BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS);
|
||||
for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++)
|
||||
acomp->aud_sample_rate[i] = 0;
|
||||
dev_priv->audio_component = acomp;
|
||||
drm_modeset_unlock_all(dev_priv->dev);
|
||||
|
||||
|
||||
Some files were not shown because too many files have changed in this diff Show More
Reference in New Issue
Block a user