Merge tag 'sound-4.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Here is the first batch of updates for sound system on 4.4-rc1.
Again at this time, the update looks fairly calm; no big changes in
either ALSA core or ASoC infrastructures, rather all small cleanups,
in addition to the new stuff as usual.
The biggest changes are about Firewire sound devices. It gained lots
of new device support, and MIDI functionality. Also there are updates
for a few still working-in-progress stuff (topology API and ASoC
skylake), too. But overall, this update should give no big surprise.
Some highlights are below:
Core:
- A few more Kconfig items for tinification; it's marked as EXPERT,
so normal user should't be bothered :)
- Refactoring with a new PCM hw_constraint helper
- Removal of unused transfer_ack_{begin,end} PCM callbacks
Firewire:
- Restructuring of code subtree, lots of refactoring
- Support AMDTP variants
- New driver for Digidesign 002/003 family
- Adds support for TASCAM FireOne to ALSA OXFW driver
- Add MIDI support to TASCAM and Digi00x devices
HD-Audio:
- Automated modalias generation for codec drivers, finally
- Improvement on heuristics for setting mixer name
- A few fixes for longstanding bugs on Creative CA0132 cards
- Addition of audio rate callback with i915 communication
- Fix suspend issue on recent Dell XPS
- Intel Lewisburg controller support
ASoC:
- Updates to the topology userspace interface
- Big updates to the Renesas support (rcar)
- More updates for supporting Intel Sky Lake systems
- New drivers for Asahi Kasei Microdevices AK4613, Allwinnner A10,
Cirrus Logic WM8998, Dialog DA7219, Nuvoton NAU8825, Rockchip
S/PDIF, and Atmel class D amplifier
USB-Audio:
- A fix for newer Roland MIDI devices
- Quirks and workarounds for Zoom R16/24 device
Misc:
- A few fixes for some old Cirrus CS46xx PCI sound boards
- Yet another fixes for some old ESS Maestro3 PCI sound boards"
* tag 'sound-4.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (330 commits)
ALSA: hda - Add Intel Lewisburg device IDs Audio
ALSA: hda - Apply pin fixup for HP ProBook 6550b
ALSA: hda - Fix lost 4k BDL boundary workaround
ALSA: maestro3: Fix Allegro mute until master volume/mute is touched
ALSA: maestro3: Enable docking support for Dell Latitude C810
ALSA: firewire-digi00x: add another rawmidi character device for MIDI control ports
ALSA: firewire-digi00x: add MIDI operations for MIDI control port
ALSA: firewire-digi00x: rename identifiers of MIDI operation for physical ports
ALSA: cs46xx: Fix suspend for all channels
ALSA: cs46xx: Fix Duplicate front for CS4294 and CS4298 codecs
ALSA: DocBook: Add soc-ops.c and soc-compress.c
ALSA: hda - Add / fix kernel doc comments
ALSA: Constify ratden/ratnum constraints
ALSA: hda - Disable 64bit address for Creative HDA controllers
ALSA: hda/realtek - Dell XPS one ALC3260 speaker no sound after resume back
ALSA: hda/ca0132 - Convert leftover pr_info() and pr_err()
ASoC: fsl: Use #ifdef instead of #if for CONFIG_PM_SLEEP
ASoC: rt5645: Sort the order for register bit defines
ASoC: dwc: add check for master/slave format
ASoC: rt5645: Add the HWEQ for the speaker output
...
diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl
index e94a10b..53f439d 100644
--- a/Documentation/DocBook/alsa-driver-api.tmpl
+++ b/Documentation/DocBook/alsa-driver-api.tmpl
@@ -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
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 84ef6a9..a27ab9f5 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -2181,10 +2181,6 @@
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 @@
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 @@
</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">
diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt
new file mode 100644
index 0000000..15a9195
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ak4613.txt
@@ -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>;
+ };
+};
diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt
index 623d4e7..340784d 100644
--- a/Documentation/devicetree/bindings/sound/ak4642.txt
+++ b/Documentation/devicetree/bindings/sound/ak4642.txt
@@ -7,7 +7,14 @@
- 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 @@
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";
+ };
+};
diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt
new file mode 100644
index 0000000..0018451
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt
@@ -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>;
+};
diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt
new file mode 100644
index 0000000..5890280
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7213.txt
@@ -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>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt
new file mode 100644
index 0000000..1b70309
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7219.txt
@@ -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>;
+ };
+ };
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index a96774c..ce55c0a 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -13,13 +13,15 @@
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"
diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt
new file mode 100644
index 0000000..d337423
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/nau8825.txt
@@ -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>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index 1173395..c57cbd6 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -4,10 +4,12 @@
- 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 @@
- 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
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
index 9b82c20..2267d24 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
@@ -12,8 +12,6 @@
- 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 @@
- 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 @@
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>;
};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
new file mode 100644
index 0000000..e64dbde
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
@@ -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>;
+};
diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt
index bac4d9a..9e62f6e 100644
--- a/Documentation/devicetree/bindings/sound/rt5640.txt
+++ b/Documentation/devicetree/bindings/sound/rt5640.txt
@@ -14,7 +14,8 @@
- 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 @@
* DMIC2
* MICBIAS1
* IN1P
- * IN1R
+ * IN1N
* IN2P
- * IN2R
+ * IN2N
+ * IN3P
+ * IN3N
* HPOL
* HPOR
* LOUTL
diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
new file mode 100644
index 0000000..c92966b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt
@@ -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";
+};
diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt
index 6a2c842..34cf70e 100644
--- a/Documentation/devicetree/bindings/sound/tdm-slot.txt
+++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt
@@ -4,11 +4,15 @@
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 @@
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.
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
deleted file mode 100644
index de8efbc..0000000
--- a/Documentation/sound/alsa/hda_codec.txt
+++ /dev/null
@@ -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.
diff --git a/MAINTAINERS b/MAINTAINERS
index 653ee9a..4e65fdf 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -3403,6 +3403,7 @@
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
diff --git a/drivers/gpu/drm/i915/i915_dma.c b/drivers/gpu/drm/i915/i915_dma.c
index ab37d11..990f656 100644
--- a/drivers/gpu/drm/i915/i915_dma.c
+++ b/drivers/gpu/drm/i915/i915_dma.c
@@ -832,6 +832,7 @@
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);
diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h
index e1db8de..22dd704 100644
--- a/drivers/gpu/drm/i915/i915_drv.h
+++ b/drivers/gpu/drm/i915/i915_drv.h
@@ -1885,6 +1885,11 @@
/* 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;
diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c
index 2a5c76f..ae8df0a 100644
--- a/drivers/gpu/drm/i915/intel_audio.c
+++ b/drivers/gpu/drm/i915/intel_audio.c
@@ -68,6 +68,31 @@
{ 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 @@
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 @@
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 @@
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 @@
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 @@
/* 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 @@
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 @@
{
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 @@
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);
diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h
index b2d56dd..89dc7d6 100644
--- a/include/drm/i915_component.h
+++ b/include/drm/i915_component.h
@@ -24,8 +24,18 @@
#ifndef _I915_COMPONENT_H_
#define _I915_COMPONENT_H_
+/* MAX_PORT is the number of port
+ * It must be sync with I915_MAX_PORTS defined i915_drv.h
+ * 5 should be enough as only HSW, BDW, SKL need such fix.
+ */
+#define MAX_PORTS 5
+
struct i915_audio_component {
struct device *dev;
+ /**
+ * @aud_sample_rate: the array of audio sample rate per port
+ */
+ int aud_sample_rate[MAX_PORTS];
const struct i915_audio_component_ops {
struct module *owner;
@@ -33,6 +43,13 @@
void (*put_power)(struct device *);
void (*codec_wake_override)(struct device *, bool enable);
int (*get_cdclk_freq)(struct device *);
+ /**
+ * @sync_audio_rate: set n/cts based on the sample rate
+ *
+ * Called from audio driver. After audio driver sets the
+ * sample rate, it will call this function to set n/cts
+ */
+ int (*sync_audio_rate)(struct device *, int port, int rate);
} *ops;
const struct i915_audio_component_audio_ops {
diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h
index 6975cbf..64f36e0 100644
--- a/include/linux/mod_devicetable.h
+++ b/include/linux/mod_devicetable.h
@@ -219,6 +219,14 @@
__u8 proto;
};
+struct hda_device_id {
+ __u32 vendor_id;
+ __u32 rev_id;
+ __u8 api_version;
+ const char *name;
+ unsigned long driver_data;
+};
+
/*
* Struct used for matching a device
*/
diff --git a/include/sound/da7213.h b/include/sound/da7213.h
index 673f5c3..e7eac897 100644
--- a/include/sound/da7213.h
+++ b/include/sound/da7213.h
@@ -44,9 +44,6 @@
enum da7213_dmic_data_sel dmic_data_sel;
enum da7213_dmic_samplephase dmic_samplephase;
enum da7213_dmic_clk_rate dmic_clk_rate;
-
- /* MCLK squaring config */
- bool mclk_squaring;
};
#endif /* _DA7213_PDATA_H */
diff --git a/include/sound/da7219-aad.h b/include/sound/da7219-aad.h
new file mode 100644
index 0000000..17802fb
--- /dev/null
+++ b/include/sound/da7219-aad.h
@@ -0,0 +1,99 @@
+/*
+ * da7219-aad.h - DA7322 ASoC Codec AAD Driver Platform Data
+ *
+ * Copyright (c) 2015 Dialog Semiconductor Ltd.
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_AAD_PDATA_H
+#define __DA7219_AAD_PDATA_H
+
+enum da7219_aad_micbias_pulse_lvl {
+ DA7219_AAD_MICBIAS_PULSE_LVL_OFF = 0,
+ DA7219_AAD_MICBIAS_PULSE_LVL_2_8V = 6,
+ DA7219_AAD_MICBIAS_PULSE_LVL_2_9V,
+};
+
+enum da7219_aad_btn_cfg {
+ DA7219_AAD_BTN_CFG_2MS = 1,
+ DA7219_AAD_BTN_CFG_5MS,
+ DA7219_AAD_BTN_CFG_10MS,
+ DA7219_AAD_BTN_CFG_50MS,
+ DA7219_AAD_BTN_CFG_100MS,
+ DA7219_AAD_BTN_CFG_200MS,
+ DA7219_AAD_BTN_CFG_500MS,
+};
+
+enum da7219_aad_mic_det_thr {
+ DA7219_AAD_MIC_DET_THR_200_OHMS = 0,
+ DA7219_AAD_MIC_DET_THR_500_OHMS,
+ DA7219_AAD_MIC_DET_THR_750_OHMS,
+ DA7219_AAD_MIC_DET_THR_1000_OHMS,
+};
+
+enum da7219_aad_jack_ins_deb {
+ DA7219_AAD_JACK_INS_DEB_5MS = 0,
+ DA7219_AAD_JACK_INS_DEB_10MS,
+ DA7219_AAD_JACK_INS_DEB_20MS,
+ DA7219_AAD_JACK_INS_DEB_50MS,
+ DA7219_AAD_JACK_INS_DEB_100MS,
+ DA7219_AAD_JACK_INS_DEB_200MS,
+ DA7219_AAD_JACK_INS_DEB_500MS,
+ DA7219_AAD_JACK_INS_DEB_1S,
+};
+
+enum da7219_aad_jack_det_rate {
+ DA7219_AAD_JACK_DET_RATE_32_64MS = 0,
+ DA7219_AAD_JACK_DET_RATE_64_128MS,
+ DA7219_AAD_JACK_DET_RATE_128_256MS,
+ DA7219_AAD_JACK_DET_RATE_256_512MS,
+};
+
+enum da7219_aad_jack_rem_deb {
+ DA7219_AAD_JACK_REM_DEB_1MS = 0,
+ DA7219_AAD_JACK_REM_DEB_5MS,
+ DA7219_AAD_JACK_REM_DEB_10MS,
+ DA7219_AAD_JACK_REM_DEB_20MS,
+};
+
+enum da7219_aad_btn_avg {
+ DA7219_AAD_BTN_AVG_1 = 0,
+ DA7219_AAD_BTN_AVG_2,
+ DA7219_AAD_BTN_AVG_4,
+ DA7219_AAD_BTN_AVG_8,
+};
+
+enum da7219_aad_adc_1bit_rpt {
+ DA7219_AAD_ADC_1BIT_RPT_1 = 0,
+ DA7219_AAD_ADC_1BIT_RPT_2,
+ DA7219_AAD_ADC_1BIT_RPT_4,
+ DA7219_AAD_ADC_1BIT_RPT_8,
+};
+
+struct da7219_aad_pdata {
+ int irq;
+
+ enum da7219_aad_micbias_pulse_lvl micbias_pulse_lvl;
+ u32 micbias_pulse_time;
+ enum da7219_aad_btn_cfg btn_cfg;
+ enum da7219_aad_mic_det_thr mic_det_thr;
+ enum da7219_aad_jack_ins_deb jack_ins_deb;
+ enum da7219_aad_jack_det_rate jack_det_rate;
+ enum da7219_aad_jack_rem_deb jack_rem_deb;
+
+ u8 a_d_btn_thr;
+ u8 d_b_btn_thr;
+ u8 b_c_btn_thr;
+ u8 c_mic_btn_thr;
+
+ enum da7219_aad_btn_avg btn_avg;
+ enum da7219_aad_adc_1bit_rpt adc_1bit_rpt;
+};
+
+#endif /* __DA7219_AAD_PDATA_H */
diff --git a/include/sound/da7219.h b/include/sound/da7219.h
new file mode 100644
index 0000000..3f39e13
--- /dev/null
+++ b/include/sound/da7219.h
@@ -0,0 +1,55 @@
+/*
+ * da7219.h - DA7219 ASoC Codec Driver Platform Data
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_PDATA_H
+#define __DA7219_PDATA_H
+
+/* LDO */
+enum da7219_ldo_lvl_sel {
+ DA7219_LDO_LVL_SEL_1_05V = 0,
+ DA7219_LDO_LVL_SEL_1_10V,
+ DA7219_LDO_LVL_SEL_1_20V,
+ DA7219_LDO_LVL_SEL_1_40V,
+};
+
+/* Mic Bias */
+enum da7219_micbias_voltage {
+ DA7219_MICBIAS_1_8V = 1,
+ DA7219_MICBIAS_2_0V,
+ DA7219_MICBIAS_2_2V,
+ DA7219_MICBIAS_2_4V,
+ DA7219_MICBIAS_2_6V,
+};
+
+/* Mic input type */
+enum da7219_mic_amp_in_sel {
+ DA7219_MIC_AMP_IN_SEL_DIFF = 0,
+ DA7219_MIC_AMP_IN_SEL_SE_P,
+ DA7219_MIC_AMP_IN_SEL_SE_N,
+};
+
+struct da7219_aad_pdata;
+
+struct da7219_pdata {
+ /* Internal LDO */
+ enum da7219_ldo_lvl_sel ldo_lvl_sel;
+
+ /* Mic */
+ enum da7219_micbias_voltage micbias_lvl;
+ enum da7219_mic_amp_in_sel mic_amp_in_sel;
+
+ /* AAD */
+ struct da7219_aad_pdata *aad_pdata;
+};
+
+#endif /* __DA7219_PDATA_H */
diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h
index 3a8fca9..8966ba7 100644
--- a/include/sound/designware_i2s.h
+++ b/include/sound/designware_i2s.h
@@ -38,6 +38,8 @@
struct i2s_platform_data {
#define DWC_I2S_PLAY (1 << 0)
#define DWC_I2S_RECORD (1 << 1)
+ #define DW_I2S_SLAVE (1 << 2)
+ #define DW_I2S_MASTER (1 << 3)
unsigned int cap;
int channel;
u32 snd_fmts;
diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h
index df70590..2767c55 100644
--- a/include/sound/hda_regmap.h
+++ b/include/sound/hda_regmap.h
@@ -67,7 +67,7 @@
* @reg: verb to write
* @val: value to write
*
- * For writing an amp value, use snd_hda_regmap_amp_update().
+ * For writing an amp value, use snd_hdac_regmap_update_amp().
*/
static inline int
snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid,
@@ -85,7 +85,7 @@
* @mask: bit mask to update
* @val: value to update
*
- * For updating an amp value, use snd_hda_regmap_amp_update().
+ * For updating an amp value, use snd_hdac_regmap_update_amp().
*/
static inline int
snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid,
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 49bc836..e2b712c 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -21,6 +21,7 @@
struct hdac_device;
struct hdac_driver;
struct hdac_widget_tree;
+struct hda_device_id;
/*
* exported bus type
@@ -28,16 +29,6 @@
extern struct bus_type snd_hda_bus_type;
/*
- * HDA device table
- */
-struct hda_device_id {
- __u32 vendor_id;
- __u32 rev_id;
- const char *name;
- unsigned long driver_data;
-};
-
-/*
* generic arrays
*/
struct snd_array {
@@ -117,6 +108,8 @@
void snd_hdac_device_exit(struct hdac_device *dev);
int snd_hdac_device_register(struct hdac_device *codec);
void snd_hdac_device_unregister(struct hdac_device *codec);
+int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name);
+int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size);
int snd_hdac_refresh_widgets(struct hdac_device *codec);
int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec);
@@ -147,6 +140,12 @@
bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid,
unsigned int format);
+int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm);
+int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm);
+bool snd_hdac_check_power_state(struct hdac_device *hdac,
+ hda_nid_t nid, unsigned int target_state);
/**
* snd_hdac_read_parm - read a codec parameter
* @codec: the codec object
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index 94210dc..a4cadd9 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -40,6 +40,13 @@
#define hbus_to_ebus(_bus) \
container_of(_bus, struct hdac_ext_bus, bus)
+#define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \
+ { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \
+ .api_version = HDA_DEV_ASOC, \
+ .driver_data = (unsigned long)(drv_data) }
+#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);
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 691e7ee..b0be092 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -265,12 +265,12 @@
struct snd_pcm_hw_constraint_ratnums {
int nrats;
- struct snd_ratnum *rats;
+ const struct snd_ratnum *rats;
};
struct snd_pcm_hw_constraint_ratdens {
int nrats;
- struct snd_ratden *rats;
+ const struct snd_ratden *rats;
};
struct snd_pcm_hw_constraint_list {
@@ -285,8 +285,6 @@
unsigned int mask;
};
-struct snd_pcm_hwptr_log;
-
/*
* userspace-provided audio timestamp config to kernel,
* structure is for internal use only and filled with dedicated unpack routine
@@ -404,10 +402,6 @@
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 */
int tstamp_type; /* timestamp type */
@@ -428,10 +422,6 @@
/* -- OSS things -- */
struct snd_pcm_oss_runtime oss;
#endif
-
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
- struct snd_pcm_hwptr_log *hwptr_log;
-#endif
};
struct snd_pcm_group { /* keep linked substreams */
@@ -980,7 +970,7 @@
int snd_interval_ranges(struct snd_interval *i, unsigned int count,
const struct snd_interval *list, unsigned int mask);
int snd_interval_ratnum(struct snd_interval *i,
- unsigned int rats_count, struct snd_ratnum *rats,
+ unsigned int rats_count, const struct snd_ratnum *rats,
unsigned int *nump, unsigned int *denp);
void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params);
@@ -1010,11 +1000,11 @@
int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var,
- struct snd_pcm_hw_constraint_ratnums *r);
+ const struct snd_pcm_hw_constraint_ratnums *r);
int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var,
- struct snd_pcm_hw_constraint_ratdens *r);
+ const struct snd_pcm_hw_constraint_ratdens *r);
int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime,
unsigned int cond,
unsigned int width,
@@ -1034,6 +1024,22 @@
snd_pcm_hw_rule_func_t func, void *private,
int dep, ...);
+/**
+ * snd_pcm_hw_constraint_single() - Constrain parameter to a single value
+ * @runtime: PCM runtime instance
+ * @var: The hw_params variable to constrain
+ * @val: The value to constrain to
+ *
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
+ */
+static inline int snd_pcm_hw_constraint_single(
+ struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
+ unsigned int val)
+{
+ return snd_pcm_hw_constraint_minmax(runtime, var, val, val);
+}
+
int snd_pcm_format_signed(snd_pcm_format_t format);
int snd_pcm_format_unsigned(snd_pcm_format_t format);
int snd_pcm_format_linear(snd_pcm_format_t format);
@@ -1117,10 +1123,16 @@
* Timer interface
*/
+#ifdef CONFIG_SND_PCM_TIMER
void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream);
void snd_pcm_timer_init(struct snd_pcm_substream *substream);
void snd_pcm_timer_done(struct snd_pcm_substream *substream);
-
+#else
+static inline void
+snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) {}
+static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {}
+static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {}
+#endif
/**
* snd_pcm_gettime - Fill the timespec depending on the timestamp mode
* @runtime: PCM runtime instance
diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h
index 56e818e..6ef629b 100644
--- a/include/sound/pxa2xx-lib.h
+++ b/include/sound/pxa2xx-lib.h
@@ -12,7 +12,6 @@
extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream);
extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream);
-extern void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id);
extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream);
extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream);
extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h
index 59d26dd..e3c84b9 100644
--- a/include/sound/rt5640.h
+++ b/include/sound/rt5640.h
@@ -12,9 +12,10 @@
#define __LINUX_SND_RT5640_H
struct rt5640_platform_data {
- /* IN1 & IN2 can optionally be differential */
+ /* IN1 & IN2 & IN3 can optionally be differential */
bool in1_diff;
bool in2_diff;
+ bool in3_diff;
bool dmic_en;
bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */
diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h
index 22734bc..a5cf615 100644
--- a/include/sound/rt5645.h
+++ b/include/sound/rt5645.h
@@ -21,6 +21,8 @@
/* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */
unsigned int jd_mode;
+ /* Invert JD when jack insert */
+ bool jd_invert;
};
#endif
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
index b9b4f28..0399352 100644
--- a/include/sound/simple_card.h
+++ b/include/sound/simple_card.h
@@ -19,6 +19,8 @@
unsigned int sysclk;
int slots;
int slot_width;
+ unsigned int tx_slot_mask;
+ unsigned int rx_slot_mask;
struct clk *clk;
};
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 2df96b1..212eaaf 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -48,10 +48,25 @@
#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
/*
- * DAI hardware signal inversions.
+ * DAI hardware signal polarity.
*
* Specifies whether the DAI can also support inverted clocks for the specified
* format.
+ *
+ * BCLK:
+ * - "normal" polarity means signal is available at rising edge of BCLK
+ * - "inverted" polarity means signal is available at falling edge of BCLK
+ *
+ * FSYNC "normal" polarity depends on the frame format:
+ * - I2S: frame consists of left then right channel data. Left channel starts
+ * with falling FSYNC edge, right channel starts with rising FSYNC edge.
+ * - Left/Right Justified: frame consists of left then right channel data.
+ * Left channel starts with rising FSYNC edge, right channel starts with
+ * falling FSYNC edge.
+ * - DSP A/B: Frame starts with rising FSYNC edge.
+ * - AC97: Frame starts with rising FSYNC edge.
+ *
+ * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
@@ -214,7 +229,7 @@
int (*suspend)(struct snd_soc_dai *dai);
int (*resume)(struct snd_soc_dai *dai);
/* compress dai */
- bool compress_dai;
+ int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
/* DAI is also used for the control bus */
bool bus_control;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 5abba03..7855cfe 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -451,6 +451,9 @@
struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
struct snd_kcontrol *kcontrol);
+struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget(
+ struct snd_kcontrol *kcontrol);
+
int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 26ede14..a8b4b9c 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -217,6 +217,13 @@
.get = xhandler_get, .put = xhandler_put, \
.private_value = \
SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) }
+#define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\
+ xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
+ xmax, xinvert) }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -226,6 +233,18 @@
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) }
+#define SOC_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_range, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, .shift = xshift, \
+ .rshift = xshift, .min = xmin, .max = xmax, \
+ .platform_max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -440,7 +459,9 @@
int snd_soc_platform_write(struct snd_soc_platform *platform,
unsigned int reg, unsigned int val);
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
-int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num);
+#ifdef CONFIG_SND_SOC_COMPRESS
+int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num);
+#endif
struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
const char *dai_link, int stream);
@@ -593,7 +614,7 @@
struct snd_ctl_elem_value *ucontrol);
int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
-int snd_soc_limit_volume(struct snd_soc_codec *codec,
+int snd_soc_limit_volume(struct snd_soc_card *card,
const char *name, int max);
int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
@@ -1603,6 +1624,8 @@
int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
const char *propname);
int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *tx_mask,
+ unsigned int *rx_mask,
unsigned int *slots,
unsigned int *slot_width);
void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card,
diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h
index 247c50b..26539a7 100644
--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@@ -83,7 +83,7 @@
#define SND_SOC_TPLG_NUM_TEXTS 16
/* ABI version */
-#define SND_SOC_TPLG_ABI_VERSION 0x3
+#define SND_SOC_TPLG_ABI_VERSION 0x4
/* Max size of TLV data */
#define SND_SOC_TPLG_TLV_SIZE 32
@@ -103,7 +103,8 @@
#define SND_SOC_TPLG_TYPE_PCM 7
#define SND_SOC_TPLG_TYPE_MANIFEST 8
#define SND_SOC_TPLG_TYPE_CODEC_LINK 9
-#define SND_SOC_TPLG_TYPE_PDATA 10
+#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
/* vendor block IDs - please add new vendor types to end */
@@ -198,7 +199,7 @@
struct snd_soc_tplg_stream_caps {
__le32 size; /* in bytes of this structure */
char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */
+ __le64 formats; /* supported formats SNDRV_PCM_FMTBIT_* */
__le32 rates; /* supported rates SNDRV_PCM_RATE_* */
__le32 rate_min; /* min rate */
__le32 rate_max; /* max rate */
@@ -217,23 +218,12 @@
*/
struct snd_soc_tplg_stream {
__le32 size; /* in bytes of this structure */
+ char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* Name of the stream */
__le64 format; /* SNDRV_PCM_FMTBIT_* */
__le32 rate; /* SNDRV_PCM_RATE_* */
__le32 period_bytes; /* size of period in bytes */
__le32 buffer_bytes; /* size of buffer in bytes */
__le32 channels; /* channels */
- __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */
- __le32 dai_fmt; /* SND_SOC_DAIFMT_ */
-} __attribute__((packed));
-
-/*
- * Duplex stream configuration supported by SW/FW.
- */
-struct snd_soc_tplg_stream_config {
- __le32 size; /* in bytes of this structure */
- char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- struct snd_soc_tplg_stream playback;
- struct snd_soc_tplg_stream capture;
} __attribute__((packed));
/*
@@ -366,11 +356,11 @@
__le32 shift; /* bits to shift */
__le32 mask; /* non-shifted mask */
__le32 subseq; /* sort within widget type */
- __u32 invert; /* invert the power bit */
- __u32 ignore_suspend; /* kept enabled over suspend */
- __u16 event_flags;
- __u16 event_type;
- __u16 num_kcontrols;
+ __le32 invert; /* invert the power bit */
+ __le32 ignore_suspend; /* kept enabled over suspend */
+ __le16 event_flags;
+ __le16 event_type;
+ __le32 num_kcontrols;
struct snd_soc_tplg_private priv;
/*
* kcontrols that relate to this widget
@@ -378,30 +368,46 @@
*/
} __attribute__((packed));
-struct snd_soc_tplg_pcm_cfg_caps {
- struct snd_soc_tplg_stream_caps caps;
- struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX];
- __le32 num_configs; /* number of configs */
-} __attribute__((packed));
/*
- * Describes SW/FW specific features of PCM or DAI link.
+ * Describes SW/FW specific features of PCM (FE DAI & DAI link).
*
- * File block representation for PCM/DAI-Link :-
+ * File block representation for PCM :-
* +-----------------------------------+-----+
* | struct snd_soc_tplg_hdr | 1 |
* +-----------------------------------+-----+
- * | struct snd_soc_tplg_dapm_pcm_dai | N |
+ * | struct snd_soc_tplg_pcm | N |
* +-----------------------------------+-----+
*/
-struct snd_soc_tplg_pcm_dai {
+struct snd_soc_tplg_pcm {
__le32 size; /* in bytes of this structure */
- char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- __le32 id; /* unique ID - used to match */
- __le32 playback; /* supports playback mode */
- __le32 capture; /* supports capture mode */
- __le32 compress; /* 1 = compressed; 0 = PCM */
- struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */
+ char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+ char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+ __le32 pcm_id; /* unique ID - used to match */
+ __le32 dai_id; /* unique ID - used to match */
+ __le32 playback; /* supports playback mode */
+ __le32 capture; /* supports capture mode */
+ __le32 compress; /* 1 = compressed; 0 = PCM */
+ struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */
+ __le32 num_streams; /* number of streams */
+ struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */
} __attribute__((packed));
+
+/*
+ * Describes the BE or CC link runtime supported configs or params
+ *
+ * File block representation for BE/CC link config :-
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_hdr | 1 |
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_link_config | N |
+ * +-----------------------------------+-----+
+ */
+struct snd_soc_tplg_link_config {
+ __le32 size; /* in bytes of this structure */
+ __le32 id; /* unique ID - used to match */
+ 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));
#endif
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index a45be6b..a82108e 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -100,9 +100,11 @@
SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */
SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */
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 */
/* Don't forget to change the following: */
- SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW
+ SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM
};
struct snd_hwdep_info {
diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h
index ec1535b..5175e16 100644
--- a/include/uapi/sound/emu10k1.h
+++ b/include/uapi/sound/emu10k1.h
@@ -34,6 +34,14 @@
#define EMU10K1_FX8010_PCM_COUNT 8
+/*
+ * Following definition is copied from linux/types.h to support compiling
+ * this header file in userspace since they are not generally available for
+ * uapi headers.
+ */
+#define __EMU10K1_DECLARE_BITMAP(name,bits) \
+ unsigned long name[(bits) / (sizeof(unsigned long) * 8)]
+
/* instruction set */
#define iMAC0 0x00 /* R = A + (X * Y >> 31) ; saturation */
#define iMAC1 0x01 /* R = A + (-X * Y >> 31) ; saturation */
@@ -300,7 +308,7 @@
struct snd_emu10k1_fx8010_code {
char name[128];
- DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */
+ __EMU10K1_DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */
__u32 __user *gpr_map; /* initializers */
unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */
@@ -313,11 +321,11 @@
unsigned int gpr_list_control_total; /* total count of GPR controls */
struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */
- DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */
+ __EMU10K1_DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */
__u32 __user *tram_data_map; /* data initializers */
__u32 __user *tram_addr_map; /* map initializers */
- DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */
+ __EMU10K1_DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */
__u32 __user *code; /* one instruction - 64 bits */
};
diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h
index 49122df..db79a12 100644
--- a/include/uapi/sound/firewire.h
+++ b/include/uapi/sound/firewire.h
@@ -9,6 +9,7 @@
#define SNDRV_FIREWIRE_EVENT_LOCK_STATUS 0x000010cc
#define SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION 0xd1ce004e
#define SNDRV_FIREWIRE_EVENT_EFW_RESPONSE 0x4e617475
+#define SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE 0x746e736c
struct snd_firewire_event_common {
unsigned int type; /* SNDRV_FIREWIRE_EVENT_xxx */
@@ -40,11 +41,17 @@
__be32 response[0]; /* some responses */
};
+struct snd_firewire_event_digi00x_message {
+ unsigned int type;
+ __u32 message; /* Digi00x-specific message */
+};
+
union snd_firewire_event {
struct snd_firewire_event_common common;
struct snd_firewire_event_lock_status lock_status;
struct snd_firewire_event_dice_notification dice_notification;
struct snd_firewire_event_efw_response efw_response;
+ struct snd_firewire_event_digi00x_message digi00x_message;
};
@@ -56,6 +63,8 @@
#define SNDRV_FIREWIRE_TYPE_FIREWORKS 2
#define SNDRV_FIREWIRE_TYPE_BEBOB 3
#define SNDRV_FIREWIRE_TYPE_OXFW 4
+#define SNDRV_FIREWIRE_TYPE_DIGI00X 5
+#define SNDRV_FIREWIRE_TYPE_TASCAM 6
/* RME, MOTU, ... */
struct snd_firewire_get_info {
diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index 5737332..c4db6f5 100644
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@@ -20,11 +20,7 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#ifdef __KERNEL__
#include <linux/types.h>
-#else
-#include <stdint.h>
-#endif
/* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */
#define HDSPM_MAX_CHANNELS 64
@@ -46,15 +42,15 @@
/* -------------------- IOCTL Peak/RMS Meters -------------------- */
struct hdspm_peak_rms {
- uint32_t input_peaks[64];
- uint32_t playback_peaks[64];
- uint32_t output_peaks[64];
+ __u32 input_peaks[64];
+ __u32 playback_peaks[64];
+ __u32 output_peaks[64];
- uint64_t input_rms[64];
- uint64_t playback_rms[64];
- uint64_t output_rms[64];
+ __u64 input_rms[64];
+ __u64 playback_rms[64];
+ __u64 output_rms[64];
- uint8_t speed; /* enum {ss, ds, qs} */
+ __u8 speed; /* enum {ss, ds, qs} */
int status2;
};
@@ -155,21 +151,21 @@
};
struct hdspm_status {
- uint8_t card_type; /* enum hdspm_io_type */
+ __u8 card_type; /* enum hdspm_io_type */
enum hdspm_syncsource autosync_source;
- uint64_t card_clock;
- uint32_t master_period;
+ __u64 card_clock;
+ __u32 master_period;
union {
struct {
- uint8_t sync_wc; /* enum hdspm_sync */
- uint8_t sync_madi; /* enum hdspm_sync */
- uint8_t sync_tco; /* enum hdspm_sync */
- uint8_t sync_in; /* enum hdspm_sync */
- uint8_t madi_input; /* enum hdspm_madi_input */
- uint8_t channel_format; /* enum hdspm_madi_channel_format */
- uint8_t frame_format; /* enum hdspm_madi_frame_format */
+ __u8 sync_wc; /* enum hdspm_sync */
+ __u8 sync_madi; /* enum hdspm_sync */
+ __u8 sync_tco; /* enum hdspm_sync */
+ __u8 sync_in; /* enum hdspm_sync */
+ __u8 madi_input; /* enum hdspm_madi_input */
+ __u8 channel_format; /* enum hdspm_madi_channel_format */
+ __u8 frame_format; /* enum hdspm_madi_frame_format */
} madi;
} card_specific;
};
@@ -184,7 +180,7 @@
#define HDSPM_ADDON_TCO 1
struct hdspm_version {
- uint8_t card_type; /* enum hdspm_io_type */
+ __u8 card_type; /* enum hdspm_io_type */
char cardname[20];
unsigned int serial;
unsigned short firmware_rev;
diff --git a/scripts/mod/devicetable-offsets.c b/scripts/mod/devicetable-offsets.c
index 5a6edac..840b973 100644
--- a/scripts/mod/devicetable-offsets.c
+++ b/scripts/mod/devicetable-offsets.c
@@ -197,5 +197,10 @@
DEVID_FIELD(ulpi_device_id, vendor);
DEVID_FIELD(ulpi_device_id, product);
+ DEVID(hda_device_id);
+ DEVID_FIELD(hda_device_id, vendor_id);
+ DEVID_FIELD(hda_device_id, rev_id);
+ DEVID_FIELD(hda_device_id, api_version);
+
return 0;
}
diff --git a/scripts/mod/file2alias.c b/scripts/mod/file2alias.c
index 9bc2cfe..5b96206 100644
--- a/scripts/mod/file2alias.c
+++ b/scripts/mod/file2alias.c
@@ -1254,6 +1254,23 @@
}
ADD_TO_DEVTABLE("ulpi", ulpi_device_id, do_ulpi_entry);
+/* Looks like: hdaudio:vNrNaN */
+static int do_hda_entry(const char *filename, void *symval, char *alias)
+{
+ DEF_FIELD(symval, hda_device_id, vendor_id);
+ DEF_FIELD(symval, hda_device_id, rev_id);
+ DEF_FIELD(symval, hda_device_id, api_version);
+
+ strcpy(alias, "hdaudio:");
+ ADD(alias, "v", vendor_id != 0, vendor_id);
+ ADD(alias, "r", rev_id != 0, rev_id);
+ ADD(alias, "a", api_version != 0, api_version);
+
+ add_wildcard(alias);
+ return 1;
+}
+ADD_TO_DEVTABLE("hdaudio", hda_device_id, do_hda_entry);
+
/* Does namelen bytes of name exactly match the symbol? */
static bool sym_is(const char *name, unsigned namelen, const char *symbol)
{
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 38590b3..fbd5dad 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -15,6 +15,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -43,7 +44,11 @@
.reset = pxa2xx_ac97_reset,
};
-static unsigned long pxa2xx_ac97_pcm_out_req = 12;
+static struct pxad_param pxa2xx_ac97_pcm_out_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 12,
+};
+
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -51,7 +56,11 @@
.filter_data = &pxa2xx_ac97_pcm_out_req,
};
-static unsigned long pxa2xx_ac97_pcm_in_req = 11;
+static struct pxad_param pxa2xx_ac97_pcm_in_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 11,
+};
+
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 01f8fdc..e9b98af 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -8,6 +8,7 @@
#include <linux/module.h>
#include <linux/dma-mapping.h>
#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -15,8 +16,6 @@
#include <sound/pxa2xx-lib.h>
#include <sound/dmaengine_pcm.h>
-#include <mach/dma.h>
-
#include "pxa2xx-pcm.h"
static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
@@ -31,7 +30,7 @@
.period_bytes_min = 32,
.period_bytes_max = 8192 - 32,
.periods_min = 1,
- .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc),
+ .periods_max = 256,
.buffer_bytes_max = 128 * 1024,
.fifo_size = 32,
};
@@ -39,65 +38,29 @@
int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *rtd = runtime->private_data;
- size_t totsize = params_buffer_bytes(params);
- size_t period = params_period_bytes(params);
- pxa_dma_desc *dma_desc;
- dma_addr_t dma_buff_phys, next_desc_phys;
- u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG;
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_dmaengine_dai_dma_data *dma_params;
+ struct dma_slave_config config;
+ int ret;
- /* temporary transition hack */
- switch (rtd->params->addr_width) {
- case DMA_SLAVE_BUSWIDTH_1_BYTE:
- dcmd |= DCMD_WIDTH1;
- break;
- case DMA_SLAVE_BUSWIDTH_2_BYTES:
- dcmd |= DCMD_WIDTH2;
- break;
- case DMA_SLAVE_BUSWIDTH_4_BYTES:
- dcmd |= DCMD_WIDTH4;
- break;
- default:
- /* can't happen */
- break;
- }
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dma_params)
+ return 0;
- switch (rtd->params->maxburst) {
- case 8:
- dcmd |= DCMD_BURST8;
- break;
- case 16:
- dcmd |= DCMD_BURST16;
- break;
- case 32:
- dcmd |= DCMD_BURST32;
- break;
- }
+ ret = snd_hwparams_to_dma_slave_config(substream, params, &config);
+ if (ret)
+ return ret;
+
+ snd_dmaengine_pcm_set_config_from_dai_data(substream,
+ snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+ &config);
+
+ ret = dmaengine_slave_config(chan, &config);
+ if (ret)
+ return ret;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- runtime->dma_bytes = totsize;
-
- dma_desc = rtd->dma_desc_array;
- next_desc_phys = rtd->dma_desc_array_phys;
- dma_buff_phys = runtime->dma_addr;
- do {
- next_desc_phys += sizeof(pxa_dma_desc);
- dma_desc->ddadr = next_desc_phys;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- dma_desc->dsadr = dma_buff_phys;
- dma_desc->dtadr = rtd->params->addr;
- } else {
- dma_desc->dsadr = rtd->params->addr;
- dma_desc->dtadr = dma_buff_phys;
- }
- if (period > totsize)
- period = totsize;
- dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN;
- dma_desc++;
- dma_buff_phys += period;
- } while (totsize -= period);
- dma_desc[-1].ddadr = rtd->dma_desc_array_phys;
return 0;
}
@@ -105,13 +68,6 @@
int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
- struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
-
- if (rtd && rtd->params && rtd->params->filter_data) {
- unsigned long req = *(unsigned long *) rtd->params->filter_data;
- DRCMR(req) = 0;
- }
-
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
}
@@ -119,100 +75,36 @@
int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
- int ret = 0;
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
- DCSR(prtd->dma_ch) = DCSR_RUN;
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- DCSR(prtd->dma_ch) &= ~DCSR_RUN;
- break;
-
- case SNDRV_PCM_TRIGGER_RESUME:
- DCSR(prtd->dma_ch) |= DCSR_RUN;
- break;
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
- DCSR(prtd->dma_ch) |= DCSR_RUN;
- break;
-
- default:
- ret = -EINVAL;
- }
-
- return ret;
+ return snd_dmaengine_pcm_trigger(substream, cmd);
}
EXPORT_SYMBOL(pxa2xx_pcm_trigger);
snd_pcm_uframes_t
pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *prtd = runtime->private_data;
-
- dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
- DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch);
- snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr);
-
- if (x == runtime->buffer_size)
- x = 0;
- return x;
+ return snd_dmaengine_pcm_pointer(substream);
}
EXPORT_SYMBOL(pxa2xx_pcm_pointer);
int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
{
- struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
- unsigned long req;
-
- if (!prtd || !prtd->params)
- return 0;
-
- if (prtd->dma_ch == -1)
- return -EINVAL;
-
- DCSR(prtd->dma_ch) &= ~DCSR_RUN;
- DCSR(prtd->dma_ch) = 0;
- DCMD(prtd->dma_ch) = 0;
- req = *(unsigned long *) prtd->params->filter_data;
- DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD;
-
return 0;
}
EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
-void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
-{
- struct snd_pcm_substream *substream = dev_id;
- int dcsr;
-
- dcsr = DCSR(dma_ch);
- DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN;
-
- if (dcsr & DCSR_ENDINTR) {
- snd_pcm_period_elapsed(substream);
- } else {
- printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n",
- dma_ch, dcsr);
- snd_pcm_stop_xrun(substream);
- }
-}
-EXPORT_SYMBOL(pxa2xx_pcm_dma_irq);
-
int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *rtd;
+ struct snd_dmaengine_dai_dma_data *dma_params;
int ret;
runtime->hw = pxa2xx_pcm_hardware;
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dma_params)
+ return 0;
+
/*
* For mysterious reasons (and despite what the manual says)
* playback samples are lost if the DMA count is not a multiple
@@ -221,48 +113,27 @@
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
if (ret)
- goto out;
+ return ret;
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
if (ret)
- goto out;
+ return ret;
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
- goto out;
+ return ret;
- ret = -ENOMEM;
- rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
- if (!rtd)
- goto out;
- rtd->dma_desc_array =
- dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE,
- &rtd->dma_desc_array_phys, GFP_KERNEL);
- if (!rtd->dma_desc_array)
- goto err1;
-
- rtd->dma_ch = -1;
- runtime->private_data = rtd;
- return 0;
-
- err1:
- kfree(rtd);
- out:
- return ret;
+ return snd_dmaengine_pcm_open_request_chan(substream,
+ pxad_filter_fn,
+ dma_params->filter_data);
}
EXPORT_SYMBOL(__pxa2xx_pcm_open);
int __pxa2xx_pcm_close(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *rtd = runtime->private_data;
-
- dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
- rtd->dma_desc_array, rtd->dma_desc_array_phys);
- kfree(rtd);
- return 0;
+ return snd_dmaengine_pcm_close_release_chan(substream);
}
EXPORT_SYMBOL(__pxa2xx_pcm_close);
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 83be8e3..83fcfac 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -46,17 +46,13 @@
rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
client->playback_params : client->capture_params;
- ret = pxa_request_dma("dma", DMA_PRIO_LOW,
- pxa2xx_pcm_dma_irq, substream);
- if (ret < 0)
- goto err2;
- rtd->dma_ch = ret;
ret = client->startup(substream);
if (!ret)
- goto out;
+ goto err2;
- pxa_free_dma(rtd->dma_ch);
+ return 0;
+
err2:
__pxa2xx_pcm_close(substream);
out:
@@ -66,9 +62,7 @@
static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
{
struct pxa2xx_pcm_client *client = substream->private_data;
- struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
- pxa_free_dma(rtd->dma_ch);
client->shutdown(substream);
return __pxa2xx_pcm_close(substream);
diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h
index 0033098..8fa2b7c 100644
--- a/sound/arm/pxa2xx-pcm.h
+++ b/sound/arm/pxa2xx-pcm.h
@@ -13,8 +13,6 @@
struct pxa2xx_runtime_data {
int dma_ch;
struct snd_dmaengine_dai_dma_data *params;
- struct pxa_dma_desc *dma_desc_array;
- dma_addr_t dma_desc_array_phys;
};
struct pxa2xx_pcm_client {
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 6c96fee..e3e9491 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -4,7 +4,7 @@
config SND_PCM
tristate
- select SND_TIMER
+ select SND_TIMER if SND_PCM_TIMER
config SND_PCM_ELD
bool
@@ -93,6 +93,17 @@
support conversion of channels, formats and rates. It will
behave like most of new OSS/Free drivers in 2.4/2.6 kernels.
+config SND_PCM_TIMER
+ bool "PCM timer interface" if EXPERT
+ default y
+ help
+ If you disable this option, pcm timer will be inavailable, so
+ those stubs used pcm timer (e.g. dmix, dsnoop & co) may work
+ incorrectlly.
+
+ For some embedded device, we may disable it to reduce memory
+ footprint, about 20KB on x86_64 platform.
+
config SND_SEQUENCER_OSS
bool "OSS Sequencer API"
depends on SND_SEQUENCER
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 3354f91..48ab4b8 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -13,8 +13,9 @@
snd-$(CONFIG_SND_VMASTER) += vmaster.o
snd-$(CONFIG_SND_JACK) += ctljack.o jack.o
-snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
+snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \
pcm_memory.o memalloc.o
+snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o
snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o
snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index a99f720..7a8c79d 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -1177,7 +1177,8 @@
struct snd_mixer_oss *mixer = entry->private_data;
char line[128], str[32], idxstr[16];
const char *cptr;
- int ch, idx;
+ unsigned int idx;
+ int ch;
struct snd_mixer_oss_assign_table *tbl;
struct slot *slot;
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 02bd969..308c9ec 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -1014,9 +1014,6 @@
snd_free_pages((void*)runtime->control,
PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)));
kfree(runtime->hw_constraints.rules);
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
- kfree(runtime->hwptr_log);
-#endif
kfree(runtime);
substream->runtime = NULL;
put_pid(substream->pid);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 7d45645..6b5a811 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -801,7 +801,7 @@
* negative error code.
*/
int snd_interval_ratnum(struct snd_interval *i,
- unsigned int rats_count, struct snd_ratnum *rats,
+ unsigned int rats_count, const struct snd_ratnum *rats,
unsigned int *nump, unsigned int *denp)
{
unsigned int best_num, best_den;
@@ -920,7 +920,8 @@
* negative error code.
*/
static int snd_interval_ratden(struct snd_interval *i,
- unsigned int rats_count, struct snd_ratden *rats,
+ unsigned int rats_count,
+ const struct snd_ratden *rats,
unsigned int *nump, unsigned int *denp)
{
unsigned int best_num, best_diff, best_den;
@@ -1339,7 +1340,7 @@
static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
- struct snd_pcm_hw_constraint_ratnums *r = rule->private;
+ const struct snd_pcm_hw_constraint_ratnums *r = rule->private;
unsigned int num = 0, den = 0;
int err;
err = snd_interval_ratnum(hw_param_interval(params, rule->var),
@@ -1363,10 +1364,10 @@
int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var,
- struct snd_pcm_hw_constraint_ratnums *r)
+ const struct snd_pcm_hw_constraint_ratnums *r)
{
return snd_pcm_hw_rule_add(runtime, cond, var,
- snd_pcm_hw_rule_ratnums, r,
+ snd_pcm_hw_rule_ratnums, (void *)r,
var, -1);
}
@@ -1375,7 +1376,7 @@
static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
- struct snd_pcm_hw_constraint_ratdens *r = rule->private;
+ const struct snd_pcm_hw_constraint_ratdens *r = rule->private;
unsigned int num = 0, den = 0;
int err = snd_interval_ratden(hw_param_interval(params, rule->var),
r->nrats, r->rats, &num, &den);
@@ -1398,10 +1399,10 @@
int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var,
- struct snd_pcm_hw_constraint_ratdens *r)
+ const struct snd_pcm_hw_constraint_ratdens *r)
{
return snd_pcm_hw_rule_add(runtime, cond, var,
- snd_pcm_hw_rule_ratdens, r,
+ snd_pcm_hw_rule_ratdens, (void *)r,
var, -1);
}
@@ -1875,20 +1876,17 @@
return;
runtime = substream->runtime;
- if (runtime->transfer_ack_begin)
- runtime->transfer_ack_begin(substream);
-
snd_pcm_stream_lock_irqsave(substream, flags);
if (!snd_pcm_running(substream) ||
snd_pcm_update_hw_ptr0(substream, 1) < 0)
goto _end;
+#ifdef CONFIG_SND_PCM_TIMER
if (substream->timer_running)
snd_timer_interrupt(substream->timer, 1);
+#endif
_end:
snd_pcm_stream_unlock_irqrestore(substream, flags);
- if (runtime->transfer_ack_end)
- runtime->transfer_ack_end(substream);
kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 75888dd..a8b27cd 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -486,6 +486,16 @@
snd_pcm_stream_unlock_irq(substream);
}
+static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream,
+ int event)
+{
+#ifdef CONFIG_SND_PCM_TIMER
+ if (substream->timer)
+ snd_timer_notify(substream->timer, event,
+ &substream->runtime->trigger_tstamp);
+#endif
+}
+
static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -650,7 +660,8 @@
}
snd_pcm_stream_unlock_irq(substream);
- if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
+ if (params->tstamp_mode < 0 ||
+ params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
return -EINVAL;
if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) &&
params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)
@@ -1042,9 +1053,7 @@
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
snd_pcm_playback_silence(substream, ULONG_MAX);
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART);
}
static struct action_ops snd_pcm_action_start = {
@@ -1092,9 +1101,7 @@
if (runtime->status->state != state) {
snd_pcm_trigger_tstamp(substream);
runtime->status->state = state;
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP);
}
wake_up(&runtime->sleep);
wake_up(&runtime->tsleep);
@@ -1208,18 +1215,12 @@
snd_pcm_trigger_tstamp(substream);
if (push) {
runtime->status->state = SNDRV_PCM_STATE_PAUSED;
- if (substream->timer)
- snd_timer_notify(substream->timer,
- SNDRV_TIMER_EVENT_MPAUSE,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE);
wake_up(&runtime->sleep);
wake_up(&runtime->tsleep);
} else {
runtime->status->state = SNDRV_PCM_STATE_RUNNING;
- if (substream->timer)
- snd_timer_notify(substream->timer,
- SNDRV_TIMER_EVENT_MCONTINUE,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE);
}
}
@@ -1267,9 +1268,7 @@
snd_pcm_trigger_tstamp(substream);
runtime->status->suspended_state = runtime->status->state;
runtime->status->state = SNDRV_PCM_STATE_SUSPENDED;
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND);
wake_up(&runtime->sleep);
wake_up(&runtime->tsleep);
}
@@ -1373,9 +1372,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_trigger_tstamp(substream);
runtime->status->state = runtime->status->suspended_state;
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME);
}
static struct action_ops snd_pcm_action_resume = {
@@ -2226,7 +2223,8 @@
snd_pcm_drop(substream);
if (substream->hw_opened) {
- if (substream->ops->hw_free != NULL)
+ if (substream->ops->hw_free &&
+ substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
substream->ops->hw_free(substream);
substream->ops->close(substream);
substream->hw_opened = 0;
diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c
index ccd8935..046cb586 100644
--- a/sound/core/seq/oss/seq_oss_readq.c
+++ b/sound/core/seq/oss/seq_oss_readq.c
@@ -91,8 +91,7 @@
q->head = q->tail = 0;
}
/* if someone sleeping, wake'em up */
- if (waitqueue_active(&q->midi_sleep))
- wake_up(&q->midi_sleep);
+ wake_up(&q->midi_sleep);
q->input_time = (unsigned long)-1;
}
@@ -138,8 +137,7 @@
q->qlen++;
/* wake up sleeper */
- if (waitqueue_active(&q->midi_sleep))
- wake_up(&q->midi_sleep);
+ wake_up(&q->midi_sleep);
spin_unlock_irqrestore(&q->lock, flags);
diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c
index d50338b..1f6788a 100644
--- a/sound/core/seq/oss/seq_oss_writeq.c
+++ b/sound/core/seq/oss/seq_oss_writeq.c
@@ -138,9 +138,7 @@
spin_lock_irqsave(&q->sync_lock, flags);
q->sync_time = time;
q->sync_event_put = 0;
- if (waitqueue_active(&q->sync_sleep)) {
- wake_up(&q->sync_sleep);
- }
+ wake_up(&q->sync_sleep);
spin_unlock_irqrestore(&q->sync_lock, flags);
}
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 8850b7d..bee0e5f 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -120,4 +120,31 @@
To compile this driver as a module, choose M here: the module
will be called snd-bebob.
+config SND_FIREWIRE_DIGI00X
+ tristate "Digidesign Digi 002/003 family support"
+ select SND_FIREWIRE_LIB
+ select SND_HWDEP
+ help
+ Say Y here to include support for Digidesign Digi 002/003 family.
+ * Digi 002 Console
+ * Digi 002 Rack
+ * Digi 003 Console
+ * Digi 003 Rack
+ * Digi 003 Rack+
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-firewire-digi00x.
+
+config SND_FIREWIRE_TASCAM
+ tristate "TASCAM FireWire series support"
+ select SND_FIREWIRE_LIB
+ select SND_HWDEP
+ help
+ Say Y here to include support for TASCAM.
+ * FW-1884
+ * FW-1082
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-firewire-tascam.
+
endif # SND_FIREWIRE
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
index 8b37f08..f5fb625 100644
--- a/sound/firewire/Makefile
+++ b/sound/firewire/Makefile
@@ -1,6 +1,5 @@
snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
- fcp.o cmp.o amdtp.o
-snd-oxfw-objs := oxfw.o
+ fcp.o cmp.o amdtp-stream.o amdtp-am824.o
snd-isight-objs := isight.o
snd-scs1x-objs := scs1x.o
@@ -11,3 +10,5 @@
obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o
obj-$(CONFIG_SND_FIREWORKS) += fireworks/
obj-$(CONFIG_SND_BEBOB) += bebob/
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += digi00x/
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += tascam/
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
new file mode 100644
index 0000000..bebddc6
--- /dev/null
+++ b/sound/firewire/amdtp-am824.c
@@ -0,0 +1,465 @@
+/*
+ * AM824 format in Audio and Music Data Transmission Protocol (IEC 61883-6)
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Copyright (c) 2015 Takashi Sakamoto <o-takashi@sakamocchi.jp>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/slab.h>
+
+#include "amdtp-am824.h"
+
+#define CIP_FMT_AM 0x10
+
+/* "Clock-based rate control mode" is just supported. */
+#define AMDTP_FDF_AM824 0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND 3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS 8
+
+struct amdtp_am824 {
+ struct snd_rawmidi_substream *midi[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+ int midi_fifo_limit;
+ int midi_fifo_used[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+ unsigned int pcm_channels;
+ unsigned int midi_ports;
+
+ u8 pcm_positions[AM824_MAX_CHANNELS_FOR_PCM];
+ u8 midi_position;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+
+ unsigned int frame_multiplier;
+};
+
+/**
+ * amdtp_am824_set_parameters - set stream parameters
+ * @s: the AMDTP stream to configure
+ * @rate: the sample rate
+ * @pcm_channels: the number of PCM samples in each data block, to be encoded
+ * as AM824 multi-bit linear audio
+ * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @double_pcm_frames: one data block transfers two PCM frames
+ *
+ * The parameters must be set before the stream is started, and must not be
+ * changed while the stream is running.
+ */
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels,
+ unsigned int midi_ports,
+ bool double_pcm_frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int midi_channels;
+ unsigned int i;
+ int err;
+
+ if (amdtp_stream_running(s))
+ return -EINVAL;
+
+ if (pcm_channels > AM824_MAX_CHANNELS_FOR_PCM)
+ return -EINVAL;
+
+ midi_channels = DIV_ROUND_UP(midi_ports, 8);
+ if (midi_channels > AM824_MAX_CHANNELS_FOR_MIDI)
+ return -EINVAL;
+
+ if (WARN_ON(amdtp_stream_running(s)) ||
+ WARN_ON(pcm_channels > AM824_MAX_CHANNELS_FOR_PCM) ||
+ WARN_ON(midi_channels > AM824_MAX_CHANNELS_FOR_MIDI))
+ return -EINVAL;
+
+ err = amdtp_stream_set_parameters(s, rate,
+ pcm_channels + midi_channels);
+ if (err < 0)
+ return err;
+
+ s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+ p->pcm_channels = pcm_channels;
+ p->midi_ports = midi_ports;
+
+ /*
+ * In IEC 61883-6, one data block represents one event. In ALSA, one
+ * event equals to one PCM frame. But Dice has a quirk at higher
+ * sampling rate to transfer two PCM frames in one data block.
+ */
+ if (double_pcm_frames)
+ p->frame_multiplier = 2;
+ else
+ p->frame_multiplier = 1;
+
+ /* init the position map for PCM and MIDI channels */
+ for (i = 0; i < pcm_channels; i++)
+ p->pcm_positions[i] = i;
+ p->midi_position = p->pcm_channels;
+
+ /*
+ * We do not know the actual MIDI FIFO size of most devices. Just
+ * assume two bytes, i.e., one byte can be received over the bus while
+ * the previous one is transmitted over MIDI.
+ * (The value here is adjusted for midi_ratelimit_per_packet().)
+ */
+ p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_parameters);
+
+/**
+ * amdtp_am824_set_pcm_position - set an index of data channel for a channel
+ * of PCM frame
+ * @s: the AMDTP stream
+ * @index: the index of data channel in an data block
+ * @position: the channel of PCM frame
+ */
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+ unsigned int position)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (index < p->pcm_channels)
+ p->pcm_positions[index] = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_position);
+
+/**
+ * amdtp_am824_set_midi_position - set a index of data channel for MIDI
+ * conformant data channel
+ * @s: the AMDTP stream
+ * @position: the index of data channel in an data block
+ */
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+ unsigned int position)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ p->midi_position = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_midi_position);
+
+static void write_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[p->pcm_positions[c]] =
+ cpu_to_be32((*src >> 8) | 0x40000000);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[p->pcm_positions[c]] =
+ cpu_to_be32((*src << 8) | 0x42000000);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[p->pcm_positions[c]]) << 8;
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int i, c, channels = p->pcm_channels;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[p->pcm_positions[c]] = cpu_to_be32(0x40000000);
+ buffer += s->data_block_quadlets;
+ }
+}
+
+/**
+ * amdtp_am824_set_pcm_format - set the PCM format
+ * @s: the AMDTP stream to configure
+ * @format: the format of the ALSA PCM device
+ *
+ * The sample format must be set after the other parameters (rate/PCM channels/
+ * MIDI) and before the stream is started, and must not be changed while the
+ * stream is running.
+ */
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_format);
+
+/**
+ * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream
+ * @s: the AMDTP stream for AM824 data block, must be initialized.
+ * @runtime: the PCM substream runtime
+ *
+ */
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ if (err < 0)
+ return err;
+
+ /* AM824 in IEC 61883-6 can deliver 24bit data. */
+ return snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_add_pcm_hw_constraints);
+
+/**
+ * amdtp_am824_midi_trigger - start/stop playback/capture with a MIDI device
+ * @s: the AMDTP stream
+ * @port: index of MIDI port
+ * @midi: the MIDI device to be started, or %NULL to stop the current device
+ *
+ * Call this function on a running isochronous stream to enable the actual
+ * transmission of MIDI data. This function should be called from the MIDI
+ * device's .trigger callback.
+ */
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (port < p->midi_ports)
+ ACCESS_ONCE(p->midi[port]) = midi;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_midi_trigger);
+
+/*
+ * To avoid sending MIDI bytes at too high a rate, assume that the receiving
+ * device has a FIFO, and track how much it is filled. This values increases
+ * by one whenever we send one byte in a packet, but the FIFO empties at
+ * a constant rate independent of our packet rate. One packet has syt_interval
+ * samples, so the number of bytes that empty out of the FIFO, per packet(!),
+ * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing
+ * fractional values, the values in midi_fifo_used[] are measured in bytes
+ * multiplied by the sample rate.
+ */
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_am824 *p = s->protocol;
+ int used;
+
+ used = p->midi_fifo_used[port];
+ if (used == 0) /* common shortcut */
+ return true;
+
+ used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+ used = max(used, 0);
+ p->midi_fifo_used[port] = used;
+
+ return used < p->midi_fifo_limit;
+}
+
+static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ p->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int f, port;
+ u8 *b;
+
+ for (f = 0; f < frames; f++) {
+ b = (u8 *)&buffer[p->midi_position];
+
+ port = (s->data_block_counter + f) % 8;
+ if (f < MAX_MIDI_RX_BLOCKS &&
+ midi_ratelimit_per_packet(s, port) &&
+ p->midi[port] != NULL &&
+ snd_rawmidi_transmit(p->midi[port], &b[1], 1) == 1) {
+ midi_rate_use_one_byte(s, port);
+ b[0] = 0x81;
+ } else {
+ b[0] = 0x80;
+ b[1] = 0;
+ }
+ b[2] = 0;
+ b[3] = 0;
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static void read_midi_messages(struct amdtp_stream *s,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int f, port;
+ int len;
+ u8 *b;
+
+ for (f = 0; f < frames; f++) {
+ port = (s->data_block_counter + f) % 8;
+ b = (u8 *)&buffer[p->midi_position];
+
+ len = b[0] - 0x80;
+ if ((1 <= len) && (len <= 3) && (p->midi[port]))
+ snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks, unsigned int *syt)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+ unsigned int pcm_frames;
+
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks * p->frame_multiplier;
+ } else {
+ write_pcm_silence(s, buffer, data_blocks);
+ pcm_frames = 0;
+ }
+
+ if (p->midi_ports)
+ write_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks, unsigned int *syt)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+ unsigned int pcm_frames;
+
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks * p->frame_multiplier;
+ } else {
+ pcm_frames = 0;
+ }
+
+ if (p->midi_ports)
+ read_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+/**
+ * amdtp_am824_init - initialize an AMDTP stream structure to handle AM824
+ * data block
+ * @s: the AMDTP stream to initialize
+ * @unit: the target of the stream
+ * @dir: the direction of stream
+ * @flags: the packet transmission method to use
+ */
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, enum cip_flags flags)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+
+ if (dir == AMDTP_IN_STREAM)
+ process_data_blocks = process_tx_data_blocks;
+ else
+ process_data_blocks = process_rx_data_blocks;
+
+ return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+ process_data_blocks,
+ sizeof(struct amdtp_am824));
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_init);
diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h
new file mode 100644
index 0000000..73b07b3
--- /dev/null
+++ b/sound/firewire/amdtp-am824.h
@@ -0,0 +1,52 @@
+#ifndef SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+#define SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+
+#include "amdtp-stream.h"
+
+#define AM824_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32
+
+#define AM824_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \
+ SNDRV_PCM_FMTBIT_S32)
+
+/*
+ * This module supports maximum 64 PCM channels for one PCM stream
+ * This is for our convenience.
+ */
+#define AM824_MAX_CHANNELS_FOR_PCM 64
+
+/*
+ * AMDTP packet can include channels for MIDI conformant data.
+ * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
+ * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
+ *
+ * This module supports maximum 1 MIDI conformant data channels.
+ * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
+ */
+#define AM824_MAX_CHANNELS_FOR_MIDI 1
+
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels,
+ unsigned int midi_ports,
+ bool double_pcm_frames);
+
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+ unsigned int position);
+
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+ unsigned int position);
+
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s,
+ snd_pcm_format_t format);
+
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi);
+
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, enum cip_flags flags);
+#endif
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp-stream.c
similarity index 70%
rename from sound/firewire/amdtp.c
rename to sound/firewire/amdtp-stream.c
index 2a153d2..ed29026 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp-stream.c
@@ -11,28 +11,14 @@
#include <linux/firewire.h>
#include <linux/module.h>
#include <linux/slab.h>
-#include <linux/sched.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/rawmidi.h>
-#include "amdtp.h"
+#include "amdtp-stream.h"
#define TICKS_PER_CYCLE 3072
#define CYCLES_PER_SECOND 8000
#define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND)
-/*
- * Nominally 3125 bytes/second, but the MIDI port's clock might be
- * 1% too slow, and the bus clock 100 ppm too fast.
- */
-#define MIDI_BYTES_PER_SECOND 3093
-
-/*
- * Several devices look only at the first eight data blocks.
- * In any case, this is more than enough for the MIDI data rate.
- */
-#define MAX_MIDI_RX_BLOCKS 8
-
#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */
/* isochronous header parameters */
@@ -55,12 +41,8 @@
#define CIP_SYT_MASK 0x0000ffff
#define CIP_SYT_NO_INFO 0xffff
-/*
- * Audio and Music transfer protocol specific parameters
- * only "Clock-based rate control mode" is supported
- */
-#define CIP_FMT_AM (0x10 << CIP_FMT_SHIFT)
-#define AMDTP_FDF_AM824 (0 << (CIP_FDF_SHIFT + 3))
+/* Audio and Music transfer protocol specific parameters */
+#define CIP_FMT_AM 0x10
#define AMDTP_FDF_NO_DATA 0xff
/* TODO: make these configurable */
@@ -78,10 +60,23 @@
* @unit: the target of the stream
* @dir: the direction of stream
* @flags: the packet transmission method to use
+ * @fmt: the value of fmt field in CIP header
+ * @process_data_blocks: callback handler to process data blocks
+ * @protocol_size: the size to allocate newly for protocol
*/
int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
- enum amdtp_stream_direction dir, enum cip_flags flags)
+ enum amdtp_stream_direction dir, enum cip_flags flags,
+ unsigned int fmt,
+ amdtp_stream_process_data_blocks_t process_data_blocks,
+ unsigned int protocol_size)
{
+ if (process_data_blocks == NULL)
+ return -EINVAL;
+
+ s->protocol = kzalloc(protocol_size, GFP_KERNEL);
+ if (!s->protocol)
+ return -ENOMEM;
+
s->unit = unit;
s->direction = dir;
s->flags = flags;
@@ -94,6 +89,9 @@
s->callbacked = false;
s->sync_slave = NULL;
+ s->fmt = fmt;
+ s->process_data_blocks = process_data_blocks;
+
return 0;
}
EXPORT_SYMBOL(amdtp_stream_init);
@@ -105,6 +103,7 @@
void amdtp_stream_destroy(struct amdtp_stream *s)
{
WARN_ON(amdtp_stream_running(s));
+ kfree(s->protocol);
mutex_destroy(&s->mutex);
}
EXPORT_SYMBOL(amdtp_stream_destroy);
@@ -141,11 +140,6 @@
{
int err;
- /* AM824 in IEC 61883-6 can deliver 24bit data */
- err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
- if (err < 0)
- goto end;
-
/*
* Currently firewire-lib processes 16 packets in one software
* interrupt callback. This equals to 2msec but actually the
@@ -190,39 +184,25 @@
* amdtp_stream_set_parameters - set stream parameters
* @s: the AMDTP stream to configure
* @rate: the sample rate
- * @pcm_channels: the number of PCM samples in each data block, to be encoded
- * as AM824 multi-bit linear audio
- * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @data_block_quadlets: the size of a data block in quadlet unit
*
* The parameters must be set before the stream is started, and must not be
* changed while the stream is running.
*/
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
- unsigned int rate,
- unsigned int pcm_channels,
- unsigned int midi_ports)
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int data_block_quadlets)
{
- unsigned int i, sfc, midi_channels;
+ unsigned int sfc;
- midi_channels = DIV_ROUND_UP(midi_ports, 8);
-
- if (WARN_ON(amdtp_stream_running(s)) |
- WARN_ON(pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM) |
- WARN_ON(midi_channels > AMDTP_MAX_CHANNELS_FOR_MIDI))
- return;
-
- for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc)
+ for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc) {
if (amdtp_rate_table[sfc] == rate)
- goto sfc_found;
- WARN_ON(1);
- return;
+ break;
+ }
+ if (sfc == ARRAY_SIZE(amdtp_rate_table))
+ return -EINVAL;
-sfc_found:
- s->pcm_channels = pcm_channels;
s->sfc = sfc;
- s->data_block_quadlets = s->pcm_channels + midi_channels;
- s->midi_ports = midi_ports;
-
+ s->data_block_quadlets = data_block_quadlets;
s->syt_interval = amdtp_syt_intervals[sfc];
/* default buffering in the device */
@@ -231,18 +211,7 @@
/* additional buffering needed to adjust for no-data packets */
s->transfer_delay += TICKS_PER_SECOND * s->syt_interval / rate;
- /* init the position map for PCM and MIDI channels */
- for (i = 0; i < pcm_channels; i++)
- s->pcm_positions[i] = i;
- s->midi_position = s->pcm_channels;
-
- /*
- * We do not know the actual MIDI FIFO size of most devices. Just
- * assume two bytes, i.e., one byte can be received over the bus while
- * the previous one is transmitted over MIDI.
- * (The value here is adjusted for midi_ratelimit_per_packet().)
- */
- s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+ return 0;
}
EXPORT_SYMBOL(amdtp_stream_set_parameters);
@@ -264,52 +233,6 @@
}
EXPORT_SYMBOL(amdtp_stream_get_max_payload);
-static void write_pcm_s16(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-static void write_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-static void read_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-
-/**
- * amdtp_stream_set_pcm_format - set the PCM format
- * @s: the AMDTP stream to configure
- * @format: the format of the ALSA PCM device
- *
- * The sample format must be set after the other parameters (rate/PCM channels/
- * MIDI) and before the stream is started, and must not be changed while the
- * stream is running.
- */
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
- snd_pcm_format_t format)
-{
- if (WARN_ON(amdtp_stream_pcm_running(s)))
- return;
-
- switch (format) {
- default:
- WARN_ON(1);
- /* fall through */
- case SNDRV_PCM_FORMAT_S16:
- if (s->direction == AMDTP_OUT_STREAM) {
- s->transfer_samples = write_pcm_s16;
- break;
- }
- WARN_ON(1);
- /* fall through */
- case SNDRV_PCM_FORMAT_S32:
- if (s->direction == AMDTP_OUT_STREAM)
- s->transfer_samples = write_pcm_s32;
- else
- s->transfer_samples = read_pcm_s32;
- break;
- }
-}
-EXPORT_SYMBOL(amdtp_stream_set_pcm_format);
-
/**
* amdtp_stream_pcm_prepare - prepare PCM device for running
* @s: the AMDTP stream
@@ -412,182 +335,12 @@
}
}
-static void write_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- const u32 *src;
-
- channels = s->pcm_channels;
- src = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- buffer[s->pcm_positions[c]] =
- cpu_to_be32((*src >> 8) | 0x40000000);
- src++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- src = (void *)runtime->dma_area;
- }
-}
-
-static void write_pcm_s16(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- const u16 *src;
-
- channels = s->pcm_channels;
- src = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- buffer[s->pcm_positions[c]] =
- cpu_to_be32((*src << 8) | 0x42000000);
- src++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- src = (void *)runtime->dma_area;
- }
-}
-
-static void read_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- u32 *dst;
-
- channels = s->pcm_channels;
- dst = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- *dst = be32_to_cpu(buffer[s->pcm_positions[c]]) << 8;
- dst++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- dst = (void *)runtime->dma_area;
- }
-}
-
-static void write_pcm_silence(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int i, c;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < s->pcm_channels; ++c)
- buffer[s->pcm_positions[c]] = cpu_to_be32(0x40000000);
- buffer += s->data_block_quadlets;
- }
-}
-
-/*
- * To avoid sending MIDI bytes at too high a rate, assume that the receiving
- * device has a FIFO, and track how much it is filled. This values increases
- * by one whenever we send one byte in a packet, but the FIFO empties at
- * a constant rate independent of our packet rate. One packet has syt_interval
- * samples, so the number of bytes that empty out of the FIFO, per packet(!),
- * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing
- * fractional values, the values in midi_fifo_used[] are measured in bytes
- * multiplied by the sample rate.
- */
-static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
-{
- int used;
-
- used = s->midi_fifo_used[port];
- if (used == 0) /* common shortcut */
- return true;
-
- used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
- used = max(used, 0);
- s->midi_fifo_used[port] = used;
-
- return used < s->midi_fifo_limit;
-}
-
-static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
-{
- s->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
-}
-
-static void write_midi_messages(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int f, port;
- u8 *b;
-
- for (f = 0; f < frames; f++) {
- b = (u8 *)&buffer[s->midi_position];
-
- port = (s->data_block_counter + f) % 8;
- if (f < MAX_MIDI_RX_BLOCKS &&
- midi_ratelimit_per_packet(s, port) &&
- s->midi[port] != NULL &&
- snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) {
- midi_rate_use_one_byte(s, port);
- b[0] = 0x81;
- } else {
- b[0] = 0x80;
- b[1] = 0;
- }
- b[2] = 0;
- b[3] = 0;
-
- buffer += s->data_block_quadlets;
- }
-}
-
-static void read_midi_messages(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int f, port;
- int len;
- u8 *b;
-
- for (f = 0; f < frames; f++) {
- port = (s->data_block_counter + f) % 8;
- b = (u8 *)&buffer[s->midi_position];
-
- len = b[0] - 0x80;
- if ((1 <= len) && (len <= 3) && (s->midi[port]))
- snd_rawmidi_receive(s->midi[port], b + 1, len);
-
- buffer += s->data_block_quadlets;
- }
-}
-
static void update_pcm_pointers(struct amdtp_stream *s,
struct snd_pcm_substream *pcm,
unsigned int frames)
{
unsigned int ptr;
- /*
- * In IEC 61883-6, one data block represents one event. In ALSA, one
- * event equals to one PCM frame. But Dice has a quirk to transfer
- * two PCM frames in one data block.
- */
- if (s->double_pcm_frames)
- frames *= 2;
-
ptr = s->pcm_buffer_pointer + frames;
if (ptr >= pcm->runtime->buffer_size)
ptr -= pcm->runtime->buffer_size;
@@ -656,23 +409,19 @@
{
__be32 *buffer;
unsigned int payload_length;
+ unsigned int pcm_frames;
struct snd_pcm_substream *pcm;
buffer = s->buffer.packets[s->packet_index].buffer;
+ pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt);
+
buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
(s->data_block_quadlets << CIP_DBS_SHIFT) |
s->data_block_counter);
- buffer[1] = cpu_to_be32(CIP_EOH | CIP_FMT_AM | AMDTP_FDF_AM824 |
- (s->sfc << CIP_FDF_SHIFT) | syt);
- buffer += 2;
-
- pcm = ACCESS_ONCE(s->pcm);
- if (pcm)
- s->transfer_samples(s, pcm, buffer, data_blocks);
- else
- write_pcm_silence(s, buffer, data_blocks);
- if (s->midi_ports)
- write_midi_messages(s, buffer, data_blocks);
+ buffer[1] = cpu_to_be32(CIP_EOH |
+ ((s->fmt << CIP_FMT_SHIFT) & CIP_FMT_MASK) |
+ ((s->fdf << CIP_FDF_SHIFT) & CIP_FDF_MASK) |
+ (syt & CIP_SYT_MASK));
s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff;
@@ -680,8 +429,9 @@
if (queue_out_packet(s, payload_length, false) < 0)
return -EIO;
- if (pcm)
- update_pcm_pointers(s, pcm, data_blocks);
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm && pcm_frames > 0)
+ update_pcm_pointers(s, pcm, pcm_frames);
/* No need to return the number of handled data blocks. */
return 0;
@@ -689,11 +439,13 @@
static int handle_in_packet(struct amdtp_stream *s,
unsigned int payload_quadlets, __be32 *buffer,
- unsigned int *data_blocks)
+ unsigned int *data_blocks, unsigned int syt)
{
u32 cip_header[2];
+ unsigned int fmt, fdf;
unsigned int data_block_quadlets, data_block_counter, dbc_interval;
- struct snd_pcm_substream *pcm = NULL;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
bool lost;
cip_header[0] = be32_to_cpu(buffer[0]);
@@ -704,19 +456,30 @@
* For convenience, also check FMT field is AM824 or not.
*/
if (((cip_header[0] & CIP_EOH_MASK) == CIP_EOH) ||
- ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH) ||
- ((cip_header[1] & CIP_FMT_MASK) != CIP_FMT_AM)) {
+ ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH)) {
dev_info_ratelimited(&s->unit->device,
"Invalid CIP header for AMDTP: %08X:%08X\n",
cip_header[0], cip_header[1]);
*data_blocks = 0;
+ pcm_frames = 0;
+ goto end;
+ }
+
+ /* Check valid protocol or not. */
+ fmt = (cip_header[1] & CIP_FMT_MASK) >> CIP_FMT_SHIFT;
+ if (fmt != s->fmt) {
+ dev_info_ratelimited(&s->unit->device,
+ "Detect unexpected protocol: %08x %08x\n",
+ cip_header[0], cip_header[1]);
+ *data_blocks = 0;
+ pcm_frames = 0;
goto end;
}
/* Calculate data blocks */
+ fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT;
if (payload_quadlets < 3 ||
- ((cip_header[1] & CIP_FDF_MASK) ==
- (AMDTP_FDF_NO_DATA << CIP_FDF_SHIFT))) {
+ (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) {
*data_blocks = 0;
} else {
data_block_quadlets =
@@ -763,16 +526,7 @@
return -EIO;
}
- if (*data_blocks > 0) {
- buffer += 2;
-
- pcm = ACCESS_ONCE(s->pcm);
- if (pcm)
- s->transfer_samples(s, pcm, buffer, *data_blocks);
-
- if (s->midi_ports)
- read_midi_messages(s, buffer, *data_blocks);
- }
+ pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt);
if (s->flags & CIP_DBC_IS_END_EVENT)
s->data_block_counter = data_block_counter;
@@ -783,8 +537,9 @@
if (queue_in_packet(s) < 0)
return -EIO;
- if (pcm)
- update_pcm_pointers(s, pcm, *data_blocks);
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm && pcm_frames > 0)
+ update_pcm_pointers(s, pcm, pcm_frames);
return 0;
}
@@ -854,15 +609,15 @@
break;
}
+ syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
if (handle_in_packet(s, payload_quadlets, buffer,
- &data_blocks) < 0) {
+ &data_blocks, syt) < 0) {
s->packet_index = -1;
break;
}
/* Process sync slave stream */
if (s->sync_slave && s->sync_slave->callbacked) {
- syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
if (handle_out_packet(s->sync_slave,
data_blocks, syt) < 0) {
s->packet_index = -1;
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp-stream.h
similarity index 77%
rename from sound/firewire/amdtp.h
rename to sound/firewire/amdtp-stream.h
index b2cf9e7..8775704 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp-stream.h
@@ -4,6 +4,7 @@
#include <linux/err.h>
#include <linux/interrupt.h>
#include <linux/mutex.h>
+#include <linux/sched.h>
#include <sound/asound.h>
#include "packets-buffer.h"
@@ -80,100 +81,78 @@
CIP_SFC_COUNT
};
-#define AMDTP_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32
-
-#define AMDTP_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \
- SNDRV_PCM_FMTBIT_S32)
-
-
-/*
- * This module supports maximum 64 PCM channels for one PCM stream
- * This is for our convenience.
- */
-#define AMDTP_MAX_CHANNELS_FOR_PCM 64
-
-/*
- * AMDTP packet can include channels for MIDI conformant data.
- * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
- * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
- *
- * This module supports maximum 1 MIDI conformant data channels.
- * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
- */
-#define AMDTP_MAX_CHANNELS_FOR_MIDI 1
-
struct fw_unit;
struct fw_iso_context;
struct snd_pcm_substream;
struct snd_pcm_runtime;
-struct snd_rawmidi_substream;
enum amdtp_stream_direction {
AMDTP_OUT_STREAM = 0,
AMDTP_IN_STREAM
};
+struct amdtp_stream;
+typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
+ struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt);
struct amdtp_stream {
struct fw_unit *unit;
enum cip_flags flags;
enum amdtp_stream_direction direction;
- struct fw_iso_context *context;
struct mutex mutex;
- enum cip_sfc sfc;
- unsigned int data_block_quadlets;
- unsigned int pcm_channels;
- unsigned int midi_ports;
- void (*transfer_samples)(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
- u8 pcm_positions[AMDTP_MAX_CHANNELS_FOR_PCM];
- u8 midi_position;
-
- unsigned int syt_interval;
- unsigned int transfer_delay;
- unsigned int source_node_id_field;
+ /* For packet processing. */
+ struct fw_iso_context *context;
struct iso_packets_buffer buffer;
-
- struct snd_pcm_substream *pcm;
- struct tasklet_struct period_tasklet;
-
int packet_index;
+
+ /* For CIP headers. */
+ unsigned int source_node_id_field;
+ unsigned int data_block_quadlets;
unsigned int data_block_counter;
-
- unsigned int data_block_state;
-
- unsigned int last_syt_offset;
- unsigned int syt_offset_state;
-
- unsigned int pcm_buffer_pointer;
- unsigned int pcm_period_pointer;
- bool pointer_flush;
- bool double_pcm_frames;
-
- struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
- int midi_fifo_limit;
- int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
-
+ unsigned int fmt;
+ unsigned int fdf;
/* quirk: fixed interval of dbc between previos/current packets. */
unsigned int tx_dbc_interval;
/* quirk: indicate the value of dbc field in a first packet. */
unsigned int tx_first_dbc;
+ /* Internal flags. */
+ enum cip_sfc sfc;
+ unsigned int syt_interval;
+ unsigned int transfer_delay;
+ unsigned int data_block_state;
+ unsigned int last_syt_offset;
+ unsigned int syt_offset_state;
+
+ /* For a PCM substream processing. */
+ struct snd_pcm_substream *pcm;
+ struct tasklet_struct period_tasklet;
+ unsigned int pcm_buffer_pointer;
+ unsigned int pcm_period_pointer;
+ bool pointer_flush;
+
+ /* To wait for first packet. */
bool callbacked;
wait_queue_head_t callback_wait;
struct amdtp_stream *sync_slave;
+
+ /* For backends to process data blocks. */
+ void *protocol;
+ amdtp_stream_process_data_blocks_t process_data_blocks;
};
int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
- enum amdtp_stream_direction dir,
- enum cip_flags flags);
+ enum amdtp_stream_direction dir, enum cip_flags flags,
+ unsigned int fmt,
+ amdtp_stream_process_data_blocks_t process_data_blocks,
+ unsigned int protocol_size);
void amdtp_stream_destroy(struct amdtp_stream *s);
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
- unsigned int rate,
- unsigned int pcm_channels,
- unsigned int midi_ports);
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int data_block_quadlets);
unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s);
int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed);
@@ -182,8 +161,7 @@
int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
struct snd_pcm_runtime *runtime);
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
- snd_pcm_format_t format);
+
void amdtp_stream_pcm_prepare(struct amdtp_stream *s);
unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s);
void amdtp_stream_pcm_abort(struct amdtp_stream *s);
@@ -240,24 +218,6 @@
ACCESS_ONCE(s->pcm) = pcm;
}
-/**
- * amdtp_stream_midi_trigger - start/stop playback/capture with a MIDI device
- * @s: the AMDTP stream
- * @port: index of MIDI port
- * @midi: the MIDI device to be started, or %NULL to stop the current device
- *
- * Call this function on a running isochronous stream to enable the actual
- * transmission of MIDI data. This function should be called from the MIDI
- * device's .trigger callback.
- */
-static inline void amdtp_stream_midi_trigger(struct amdtp_stream *s,
- unsigned int port,
- struct snd_rawmidi_substream *midi)
-{
- if (port < s->midi_ports)
- ACCESS_ONCE(s->midi[port]) = midi;
-}
-
static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
{
return sfc & 1;
diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile
index 6cf470c..af7ed66 100644
--- a/sound/firewire/bebob/Makefile
+++ b/sound/firewire/bebob/Makefile
@@ -1,4 +1,4 @@
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
-obj-m += snd-bebob.o
+obj-$(CONFIG_SND_BEBOB) += snd-bebob.o
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 27a04ac..091290d 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -41,7 +41,8 @@
#define VEN_EDIROL 0x000040ab
#define VEN_PRESONUS 0x00000a92
#define VEN_BRIDGECO 0x000007f5
-#define VEN_MACKIE 0x0000000f
+#define VEN_MACKIE1 0x0000000f
+#define VEN_MACKIE2 0x00000ff2
#define VEN_STANTON 0x00001260
#define VEN_TASCAM 0x0000022e
#define VEN_BEHRINGER 0x00001564
@@ -334,7 +335,7 @@
snd_card_free_when_closed(bebob->card);
}
-static struct snd_bebob_rate_spec normal_rate_spec = {
+static const struct snd_bebob_rate_spec normal_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate
};
@@ -360,9 +361,9 @@
/* BridgeCo, Audio5 */
SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal),
/* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */
- SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010065, &spec_normal),
+ SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal),
/* Mackie, d.2 (Firewire Option) */
- SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010067, &spec_normal),
+ SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal),
/* Stanton, ScratchAmp */
SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal),
/* Tascam, IF-FW DM */
diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index d23caca..4d8fcc7 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -31,7 +31,7 @@
#include "../fcp.h"
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
/* basic register addresses on DM1000/DM1100/DM1500 */
@@ -70,9 +70,9 @@
int (*get)(struct snd_bebob *bebob, u32 *target, unsigned int size);
};
struct snd_bebob_spec {
- struct snd_bebob_clock_spec *clock;
- struct snd_bebob_rate_spec *rate;
- struct snd_bebob_meter_spec *meter;
+ const struct snd_bebob_clock_spec *clock;
+ const struct snd_bebob_rate_spec *rate;
+ const struct snd_bebob_meter_spec *meter;
};
struct snd_bebob {
@@ -235,19 +235,19 @@
int snd_bebob_create_hwdep_device(struct snd_bebob *bebob);
/* model specific operations */
-extern struct snd_bebob_spec phase88_rack_spec;
-extern struct snd_bebob_spec phase24_series_spec;
-extern struct snd_bebob_spec yamaha_go_spec;
-extern struct snd_bebob_spec saffirepro_26_spec;
-extern struct snd_bebob_spec saffirepro_10_spec;
-extern struct snd_bebob_spec saffire_le_spec;
-extern struct snd_bebob_spec saffire_spec;
-extern struct snd_bebob_spec maudio_fw410_spec;
-extern struct snd_bebob_spec maudio_audiophile_spec;
-extern struct snd_bebob_spec maudio_solo_spec;
-extern struct snd_bebob_spec maudio_ozonic_spec;
-extern struct snd_bebob_spec maudio_nrv10_spec;
-extern struct snd_bebob_spec maudio_special_spec;
+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 saffirepro_26_spec;
+extern const struct snd_bebob_spec saffirepro_10_spec;
+extern const struct snd_bebob_spec saffire_le_spec;
+extern const struct snd_bebob_spec saffire_spec;
+extern const struct snd_bebob_spec maudio_fw410_spec;
+extern const struct snd_bebob_spec maudio_audiophile_spec;
+extern const struct snd_bebob_spec maudio_solo_spec;
+extern const struct snd_bebob_spec maudio_ozonic_spec;
+extern const struct snd_bebob_spec maudio_nrv10_spec;
+extern const struct snd_bebob_spec maudio_special_spec;
int snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814);
int snd_bebob_maudio_load_firmware(struct fw_unit *unit);
diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c
index a1a3949..f110900 100644
--- a/sound/firewire/bebob/bebob_focusrite.c
+++ b/sound/firewire/bebob/bebob_focusrite.c
@@ -200,7 +200,7 @@
return err;
}
-struct snd_bebob_spec saffire_le_spec;
+const struct snd_bebob_spec saffire_le_spec;
static enum snd_bebob_clock_type saffire_both_clk_src_types[] = {
SND_BEBOB_CLOCK_TYPE_INTERNAL,
SND_BEBOB_CLOCK_TYPE_EXTERNAL,
@@ -229,7 +229,7 @@
static int
saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
{
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
unsigned int channels;
u64 offset;
int err;
@@ -260,60 +260,60 @@
return err;
}
-static struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
.get = &saffirepro_both_clk_freq_get,
.set = &saffirepro_both_clk_freq_set,
};
/* Saffire Pro 26 I/O */
-static struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
.num = ARRAY_SIZE(saffirepro_26_clk_src_types),
.types = saffirepro_26_clk_src_types,
.get = &saffirepro_both_clk_src_get,
};
-struct snd_bebob_spec saffirepro_26_spec = {
+const struct snd_bebob_spec saffirepro_26_spec = {
.clock = &saffirepro_26_clk_spec,
.rate = &saffirepro_both_rate_spec,
.meter = NULL
};
/* Saffire Pro 10 I/O */
-static struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
.num = ARRAY_SIZE(saffirepro_10_clk_src_types),
.types = saffirepro_10_clk_src_types,
.get = &saffirepro_both_clk_src_get,
};
-struct snd_bebob_spec saffirepro_10_spec = {
+const struct snd_bebob_spec saffirepro_10_spec = {
.clock = &saffirepro_10_clk_spec,
.rate = &saffirepro_both_rate_spec,
.meter = NULL
};
-static struct snd_bebob_rate_spec saffire_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffire_both_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-static struct snd_bebob_clock_spec saffire_both_clk_spec = {
+static const struct snd_bebob_clock_spec saffire_both_clk_spec = {
.num = ARRAY_SIZE(saffire_both_clk_src_types),
.types = saffire_both_clk_src_types,
.get = &saffire_both_clk_src_get,
};
/* Saffire LE */
-static struct snd_bebob_meter_spec saffire_le_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_le_meter_spec = {
.num = ARRAY_SIZE(saffire_le_meter_labels),
.labels = saffire_le_meter_labels,
.get = &saffire_meter_get,
};
-struct snd_bebob_spec saffire_le_spec = {
+const struct snd_bebob_spec saffire_le_spec = {
.clock = &saffire_both_clk_spec,
.rate = &saffire_both_rate_spec,
.meter = &saffire_le_meter_spec
};
/* Saffire */
-static struct snd_bebob_meter_spec saffire_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_meter_spec = {
.num = ARRAY_SIZE(saffire_meter_labels),
.labels = saffire_meter_labels,
.get = &saffire_meter_get,
};
-struct snd_bebob_spec saffire_spec = {
+const struct snd_bebob_spec saffire_spec = {
.clock = &saffire_both_clk_spec,
.rate = &saffire_both_rate_spec,
.meter = &saffire_meter_spec
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index 057495d..07e5abd 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -628,7 +628,7 @@
static int
special_meter_get(struct snd_bebob *bebob, u32 *target, unsigned int size)
{
- u16 *buf;
+ __be16 *buf;
unsigned int i, c, channels;
int err;
@@ -687,7 +687,7 @@
static int
normal_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
{
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
unsigned int c, channels;
int err;
@@ -712,85 +712,85 @@
}
/* for special customized devices */
-static struct snd_bebob_rate_spec special_rate_spec = {
+static const struct snd_bebob_rate_spec special_rate_spec = {
.get = &special_get_rate,
.set = &special_set_rate,
};
-static struct snd_bebob_clock_spec special_clk_spec = {
+static const struct snd_bebob_clock_spec special_clk_spec = {
.num = ARRAY_SIZE(special_clk_types),
.types = special_clk_types,
.get = &special_clk_get,
};
-static struct snd_bebob_meter_spec special_meter_spec = {
+static const struct snd_bebob_meter_spec special_meter_spec = {
.num = ARRAY_SIZE(special_meter_labels),
.labels = special_meter_labels,
.get = &special_meter_get
};
-struct snd_bebob_spec maudio_special_spec = {
+const struct snd_bebob_spec maudio_special_spec = {
.clock = &special_clk_spec,
.rate = &special_rate_spec,
.meter = &special_meter_spec
};
/* Firewire 410 specification */
-static struct snd_bebob_rate_spec usual_rate_spec = {
+static const struct snd_bebob_rate_spec usual_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-static struct snd_bebob_meter_spec fw410_meter_spec = {
+static const struct snd_bebob_meter_spec fw410_meter_spec = {
.num = ARRAY_SIZE(fw410_meter_labels),
.labels = fw410_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_fw410_spec = {
+const struct snd_bebob_spec maudio_fw410_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &fw410_meter_spec
};
/* Firewire Audiophile specification */
-static struct snd_bebob_meter_spec audiophile_meter_spec = {
+static const struct snd_bebob_meter_spec audiophile_meter_spec = {
.num = ARRAY_SIZE(audiophile_meter_labels),
.labels = audiophile_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_audiophile_spec = {
+const struct snd_bebob_spec maudio_audiophile_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &audiophile_meter_spec
};
/* Firewire Solo specification */
-static struct snd_bebob_meter_spec solo_meter_spec = {
+static const struct snd_bebob_meter_spec solo_meter_spec = {
.num = ARRAY_SIZE(solo_meter_labels),
.labels = solo_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_solo_spec = {
+const struct snd_bebob_spec maudio_solo_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &solo_meter_spec
};
/* Ozonic specification */
-static struct snd_bebob_meter_spec ozonic_meter_spec = {
+static const struct snd_bebob_meter_spec ozonic_meter_spec = {
.num = ARRAY_SIZE(ozonic_meter_labels),
.labels = ozonic_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_ozonic_spec = {
+const struct snd_bebob_spec maudio_ozonic_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &ozonic_meter_spec
};
/* NRV10 specification */
-static struct snd_bebob_meter_spec nrv10_meter_spec = {
+static const struct snd_bebob_meter_spec nrv10_meter_spec = {
.num = ARRAY_SIZE(nrv10_meter_labels),
.labels = nrv10_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_nrv10_spec = {
+const struct snd_bebob_spec maudio_nrv10_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &nrv10_meter_spec
diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c
index 5681143..90d95be 100644
--- a/sound/firewire/bebob/bebob_midi.c
+++ b/sound/firewire/bebob/bebob_midi.c
@@ -72,11 +72,11 @@
spin_lock_irqsave(&bebob->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&bebob->tx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&bebob->tx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&bebob->tx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&bebob->tx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&bebob->lock, flags);
}
@@ -89,11 +89,11 @@
spin_lock_irqsave(&bebob->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&bebob->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&bebob->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&bebob->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&bebob->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&bebob->lock, flags);
}
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index c0f018a..ef224d6 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -122,11 +122,11 @@
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
s = &bebob->tx_stream;
formations = bebob->tx_stream_formations;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
s = &bebob->rx_stream;
formations = bebob->rx_stream_formations;
}
@@ -146,7 +146,7 @@
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
end:
return err;
}
@@ -155,7 +155,7 @@
pcm_open(struct snd_pcm_substream *substream)
{
struct snd_bebob *bebob = substream->private_data;
- struct snd_bebob_rate_spec *spec = bebob->spec->rate;
+ const struct snd_bebob_rate_spec *spec = bebob->spec->rate;
unsigned int sampling_rate;
enum snd_bebob_clock_type src;
int err;
@@ -220,8 +220,8 @@
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&bebob->substreams_counter);
- amdtp_stream_set_pcm_format(&bebob->tx_stream,
- params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&bebob->tx_stream, params_format(hw_params));
return 0;
}
@@ -239,8 +239,8 @@
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&bebob->substreams_counter);
- amdtp_stream_set_pcm_format(&bebob->rx_stream,
- params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&bebob->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c
index 301cc6a..ec24f96 100644
--- a/sound/firewire/bebob/bebob_proc.c
+++ b/sound/firewire/bebob/bebob_proc.c
@@ -73,7 +73,7 @@
struct snd_info_buffer *buffer)
{
struct snd_bebob *bebob = entry->private_data;
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
u32 *buf;
unsigned int i, c, channels, size;
@@ -138,8 +138,8 @@
"SYT-Match",
};
struct snd_bebob *bebob = entry->private_data;
- struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
enum snd_bebob_clock_type src;
unsigned int rate;
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 5be5242..926e5dc 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -119,7 +119,7 @@
int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob,
enum snd_bebob_clock_type *src)
{
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7];
unsigned int id;
enum avc_bridgeco_plug_type type;
@@ -338,7 +338,7 @@
err = -ENOSYS;
goto end;
}
- s->midi_position = stm_pos;
+ amdtp_am824_set_midi_position(s, stm_pos);
midi = stm_pos;
break;
/* for PCM data channel */
@@ -354,11 +354,12 @@
case 0x09: /* Digital */
default:
location = pcm + sec_loc;
- if (location >= AMDTP_MAX_CHANNELS_FOR_PCM) {
+ if (location >= AM824_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
- s->pcm_positions[location] = stm_pos;
+ amdtp_am824_set_pcm_position(s, location,
+ stm_pos);
break;
}
}
@@ -427,12 +428,19 @@
index = get_formation_index(rate);
pcm_channels = bebob->tx_stream_formations[index].pcm;
midi_channels = bebob->tx_stream_formations[index].midi;
- amdtp_stream_set_parameters(&bebob->tx_stream,
- rate, pcm_channels, midi_channels * 8);
+ err = amdtp_am824_set_parameters(&bebob->tx_stream, rate,
+ pcm_channels, midi_channels * 8,
+ false);
+ if (err < 0)
+ goto end;
+
pcm_channels = bebob->rx_stream_formations[index].pcm;
midi_channels = bebob->rx_stream_formations[index].midi;
- amdtp_stream_set_parameters(&bebob->rx_stream,
- rate, pcm_channels, midi_channels * 8);
+ err = amdtp_am824_set_parameters(&bebob->rx_stream, rate,
+ pcm_channels, midi_channels * 8,
+ false);
+ if (err < 0)
+ goto end;
/* establish connections for both streams */
err = cmp_connection_establish(&bebob->out_conn,
@@ -530,8 +538,8 @@
if (err < 0)
goto end;
- err = amdtp_stream_init(&bebob->tx_stream, bebob->unit,
- AMDTP_IN_STREAM, CIP_BLOCKING);
+ err = amdtp_am824_init(&bebob->tx_stream, bebob->unit,
+ AMDTP_IN_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
destroy_both_connections(bebob);
@@ -559,8 +567,8 @@
if (bebob->maudio_special_quirk)
bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC;
- err = amdtp_stream_init(&bebob->rx_stream, bebob->unit,
- AMDTP_OUT_STREAM, CIP_BLOCKING);
+ err = amdtp_am824_init(&bebob->rx_stream, bebob->unit,
+ AMDTP_OUT_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
amdtp_stream_destroy(&bebob->rx_stream);
@@ -572,7 +580,7 @@
int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
{
- struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+ const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
struct amdtp_stream *master, *slave;
enum cip_flags sync_mode;
unsigned int curr_rate;
@@ -864,8 +872,8 @@
}
}
- if (formation[i].pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
- formation[i].midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+ if (formation[i].pcm > AM824_MAX_CHANNELS_FOR_PCM ||
+ formation[i].midi > AM824_MAX_CHANNELS_FOR_MIDI)
return -ENOSYS;
return 0;
@@ -959,7 +967,7 @@
int snd_bebob_stream_discover(struct snd_bebob *bebob)
{
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES];
enum avc_bridgeco_plug_type type;
unsigned int i;
diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c
index 9242e33..c38358b 100644
--- a/sound/firewire/bebob/bebob_terratec.c
+++ b/sound/firewire/bebob/bebob_terratec.c
@@ -55,30 +55,30 @@
return 0;
}
-static struct snd_bebob_rate_spec phase_series_rate_spec = {
+static const struct snd_bebob_rate_spec phase_series_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
/* PHASE 88 Rack FW */
-static struct snd_bebob_clock_spec phase88_rack_clk = {
+static const struct snd_bebob_clock_spec phase88_rack_clk = {
.num = ARRAY_SIZE(phase88_rack_clk_src_types),
.types = phase88_rack_clk_src_types,
.get = &phase88_rack_clk_src_get,
};
-struct snd_bebob_spec phase88_rack_spec = {
+const struct snd_bebob_spec phase88_rack_spec = {
.clock = &phase88_rack_clk,
.rate = &phase_series_rate_spec,
.meter = NULL
};
/* 'PHASE 24 FW' and 'PHASE X24 FW' */
-static struct snd_bebob_clock_spec phase24_series_clk = {
+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,
};
-struct snd_bebob_spec phase24_series_spec = {
+const struct snd_bebob_spec phase24_series_spec = {
.clock = &phase24_series_clk,
.rate = &phase_series_rate_spec,
.meter = NULL
diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c
index 5810170..90d4404 100644
--- a/sound/firewire/bebob/bebob_yamaha.c
+++ b/sound/firewire/bebob/bebob_yamaha.c
@@ -46,16 +46,16 @@
return 0;
}
-static struct snd_bebob_clock_spec clock_spec = {
+static const struct snd_bebob_clock_spec clock_spec = {
.num = ARRAY_SIZE(clk_src_types),
.types = clk_src_types,
.get = &clk_src_get,
};
-static struct snd_bebob_rate_spec rate_spec = {
+static const struct snd_bebob_rate_spec rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-struct snd_bebob_spec yamaha_go_spec = {
+const struct snd_bebob_spec yamaha_go_spec = {
.clock = &clock_spec,
.rate = &rate_spec,
.meter = NULL
diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile
index 9ef228e..55b4be9 100644
--- a/sound/firewire/dice/Makefile
+++ b/sound/firewire/dice/Makefile
@@ -1,3 +1,3 @@
snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
dice-pcm.o dice-hwdep.o dice.o
-obj-m += snd-dice.o
+obj-$(CONFIG_SND_DICE) += snd-dice.o
diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c
index fe43ce7..151b09f 100644
--- a/sound/firewire/dice/dice-midi.c
+++ b/sound/firewire/dice/dice-midi.c
@@ -52,10 +52,10 @@
spin_lock_irqsave(&dice->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&dice->tx_stream,
+ amdtp_am824_midi_trigger(&dice->tx_stream,
substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&dice->tx_stream,
+ amdtp_am824_midi_trigger(&dice->tx_stream,
substrm->number, NULL);
spin_unlock_irqrestore(&dice->lock, flags);
@@ -69,11 +69,11 @@
spin_lock_irqsave(&dice->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&dice->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&dice->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&dice->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&dice->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&dice->lock, flags);
}
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 4e67b1d..9b34319 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -133,11 +133,11 @@
SNDRV_PCM_INFO_BLOCK_TRANSFER;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- hw->formats = AMDTP_IN_PCM_FORMAT_BITS;
+ hw->formats = AM824_IN_PCM_FORMAT_BITS;
stream = &dice->tx_stream;
pcm_channels = dice->tx_channels;
} else {
- hw->formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ hw->formats = AM824_OUT_PCM_FORMAT_BITS;
stream = &dice->rx_stream;
pcm_channels = dice->rx_channels;
}
@@ -156,7 +156,7 @@
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
end:
return err;
}
@@ -243,8 +243,7 @@
mutex_unlock(&dice->mutex);
}
- amdtp_stream_set_pcm_format(&dice->tx_stream,
- params_format(hw_params));
+ amdtp_am824_set_pcm_format(&dice->tx_stream, params_format(hw_params));
return 0;
}
@@ -265,8 +264,7 @@
mutex_unlock(&dice->mutex);
}
- amdtp_stream_set_pcm_format(&dice->rx_stream,
- params_format(hw_params));
+ amdtp_am824_set_pcm_format(&dice->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 07dbd01..a6a39f7 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -44,16 +44,16 @@
static void release_resources(struct snd_dice *dice,
struct fw_iso_resources *resources)
{
- unsigned int channel;
+ __be32 channel;
/* Reset channel number */
channel = cpu_to_be32((u32)-1);
if (resources == &dice->tx_resources)
snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
else
snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
fw_iso_resources_free(resources);
}
@@ -62,7 +62,7 @@
struct fw_iso_resources *resources,
unsigned int max_payload_bytes)
{
- unsigned int channel;
+ __be32 channel;
int err;
err = fw_iso_resources_allocate(resources, max_payload_bytes,
@@ -74,10 +74,10 @@
channel = cpu_to_be32(resources->channel);
if (resources == &dice->tx_resources)
err = snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
else
err = snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
if (err < 0)
release_resources(dice, resources);
end:
@@ -100,6 +100,7 @@
{
struct fw_iso_resources *resources;
unsigned int i, mode, pcm_chs, midi_ports;
+ bool double_pcm_frames;
int err;
err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
@@ -125,21 +126,24 @@
* For this quirk, blocking mode is required and PCM buffer size should
* be aligned to SYT_INTERVAL.
*/
- if (mode > 1) {
+ double_pcm_frames = mode > 1;
+ if (double_pcm_frames) {
rate /= 2;
pcm_chs *= 2;
- stream->double_pcm_frames = true;
- } else {
- stream->double_pcm_frames = false;
}
- amdtp_stream_set_parameters(stream, rate, pcm_chs, midi_ports);
- if (mode > 1) {
+ err = amdtp_am824_set_parameters(stream, rate, pcm_chs, midi_ports,
+ double_pcm_frames);
+ if (err < 0)
+ goto end;
+
+ if (double_pcm_frames) {
pcm_chs /= 2;
for (i = 0; i < pcm_chs; i++) {
- stream->pcm_positions[i] = i * 2;
- stream->pcm_positions[i + pcm_chs] = i * 2 + 1;
+ amdtp_am824_set_pcm_position(stream, i, i * 2);
+ amdtp_am824_set_pcm_position(stream, i + pcm_chs,
+ i * 2 + 1);
}
}
@@ -302,7 +306,7 @@
goto end;
resources->channels_mask = 0x00000000ffffffffuLL;
- err = amdtp_stream_init(stream, dice->unit, dir, CIP_BLOCKING);
+ err = amdtp_am824_init(stream, dice->unit, dir, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
fw_iso_resources_destroy(resources);
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 70a111d7f..5d99436 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -29,7 +29,8 @@
struct fw_csr_iterator it;
int key, val, vendor = -1, model = -1, err;
unsigned int category, i;
- __be32 *pointers, value;
+ __be32 *pointers;
+ u32 value;
__be32 version;
pointers = kmalloc_array(ARRAY_SIZE(min_values), sizeof(__be32),
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index ecf5dc8..101550ac 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -34,7 +34,7 @@
#include <sound/pcm_params.h>
#include <sound/rawmidi.h>
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../iso-resources.h"
#include "../lib.h"
#include "dice-interface.h"
diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile
new file mode 100644
index 0000000..1123e68
--- /dev/null
+++ b/sound/firewire/digi00x/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \
+ digi00x-pcm.o digi00x-hwdep.o \
+ digi00x-transaction.o digi00x-midi.o digi00x.o
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
new file mode 100644
index 0000000..b02a5e8c
--- /dev/null
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -0,0 +1,442 @@
+/*
+ * amdtp-dot.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ * Copyright (C) 2012 Robin Gareus <robin@gareus.org>
+ * Copyright (C) 2012 Damien Zammit <damien@zamaudio.com>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "digi00x.h"
+
+#define CIP_FMT_AM 0x10
+
+/* 'Clock-based rate control mode' is just supported. */
+#define AMDTP_FDF_AM824 0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND 3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS 8
+
+/*
+ * The double-oh-three algorithm was discovered by Robin Gareus and Damien
+ * Zammit in 2012, with reverse-engineering for Digi 003 Rack.
+ */
+struct dot_state {
+ u8 carry;
+ u8 idx;
+ unsigned int off;
+};
+
+struct amdtp_dot {
+ unsigned int pcm_channels;
+ struct dot_state state;
+
+ unsigned int midi_ports;
+ /* 2 = MAX(DOT_MIDI_IN_PORTS, DOT_MIDI_OUT_PORTS) */
+ struct snd_rawmidi_substream *midi[2];
+ int midi_fifo_used[2];
+ int midi_fifo_limit;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+};
+
+/*
+ * double-oh-three look up table
+ *
+ * @param idx index byte (audio-sample data) 0x00..0xff
+ * @param off channel offset shift
+ * @return salt to XOR with given data
+ */
+#define BYTE_PER_SAMPLE (4)
+#define MAGIC_DOT_BYTE (2)
+#define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE)
+static const u8 dot_scrt(const u8 idx, const unsigned int off)
+{
+ /*
+ * the length of the added pattern only depends on the lower nibble
+ * of the last non-zero data
+ */
+ static const u8 len[16] = {0, 1, 3, 5, 7, 9, 11, 13, 14,
+ 12, 10, 8, 6, 4, 2, 0};
+
+ /*
+ * the lower nibble of the salt. Interleaved sequence.
+ * this is walked backwards according to len[]
+ */
+ static const u8 nib[15] = {0x8, 0x7, 0x9, 0x6, 0xa, 0x5, 0xb, 0x4,
+ 0xc, 0x3, 0xd, 0x2, 0xe, 0x1, 0xf};
+
+ /* circular list for the salt's hi nibble. */
+ static const u8 hir[15] = {0x0, 0x6, 0xf, 0x8, 0x7, 0x5, 0x3, 0x4,
+ 0xc, 0xd, 0xe, 0x1, 0x2, 0xb, 0xa};
+
+ /*
+ * start offset for upper nibble mapping.
+ * note: 9 is /special/. In the case where the high nibble == 0x9,
+ * hir[] is not used and - coincidentally - the salt's hi nibble is
+ * 0x09 regardless of the offset.
+ */
+ static const u8 hio[16] = {0, 11, 12, 6, 7, 5, 1, 4,
+ 3, 0x00, 14, 13, 8, 9, 10, 2};
+
+ const u8 ln = idx & 0xf;
+ const u8 hn = (idx >> 4) & 0xf;
+ const u8 hr = (hn == 0x9) ? 0x9 : hir[(hio[hn] + off) % 15];
+
+ if (len[ln] < off)
+ return 0x00;
+
+ return ((nib[14 + off - len[ln]]) | (hr << 4));
+}
+
+static void dot_encode_step(struct dot_state *state, __be32 *const buffer)
+{
+ u8 * const data = (u8 *) buffer;
+
+ if (data[MAGIC_DOT_BYTE] != 0x00) {
+ state->off = 0;
+ state->idx = data[MAGIC_DOT_BYTE] ^ state->carry;
+ }
+ data[MAGIC_DOT_BYTE] ^= state->carry;
+ state->carry = dot_scrt(state->idx, ++(state->off));
+}
+
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels)
+{
+ struct amdtp_dot *p = s->protocol;
+ int err;
+
+ if (amdtp_stream_running(s))
+ return -EBUSY;
+
+ /*
+ * A first data channel is for MIDI conformant data channel, the rest is
+ * Multi Bit Linear Audio data channel.
+ */
+ err = amdtp_stream_set_parameters(s, rate, pcm_channels + 1);
+ if (err < 0)
+ return err;
+
+ s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+ p->pcm_channels = pcm_channels;
+
+ if (s->direction == AMDTP_IN_STREAM)
+ p->midi_ports = DOT_MIDI_IN_PORTS;
+ else
+ p->midi_ports = DOT_MIDI_OUT_PORTS;
+
+ /*
+ * We do not know the actual MIDI FIFO size of most devices. Just
+ * assume two bytes, i.e., one byte can be received over the bus while
+ * the previous one is transmitted over MIDI.
+ * (The value here is adjusted for midi_ratelimit_per_packet().)
+ */
+ p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+ return 0;
+}
+
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32((*src >> 8) | 0x40000000);
+ dot_encode_step(&p->state, &buffer[c]);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32((*src << 8) | 0x40000000);
+ dot_encode_step(&p->state, &buffer[c]);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[c]) << 8;
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int channels, i, c;
+
+ channels = p->pcm_channels;
+
+ buffer++;
+ for (i = 0; i < data_blocks; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[c] = cpu_to_be32(0x40000000);
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_dot *p = s->protocol;
+ int used;
+
+ used = p->midi_fifo_used[port];
+ if (used == 0)
+ return true;
+
+ used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+ used = max(used, 0);
+ p->midi_fifo_used[port] = used;
+
+ return used < p->midi_fifo_limit;
+}
+
+static inline void midi_use_bytes(struct amdtp_stream *s,
+ unsigned int port, unsigned int count)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ p->midi_fifo_used[port] += amdtp_rate_table[s->sfc] * count;
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int f, port;
+ int len;
+ u8 *b;
+
+ for (f = 0; f < data_blocks; f++) {
+ port = (s->data_block_counter + f) % 8;
+ b = (u8 *)&buffer[0];
+
+ len = 0;
+ if (port < p->midi_ports &&
+ midi_ratelimit_per_packet(s, port) &&
+ p->midi[port] != NULL)
+ len = snd_rawmidi_transmit(p->midi[port], b + 1, 2);
+
+ if (len > 0) {
+ b[3] = (0x10 << port) | len;
+ midi_use_bytes(s, port, len);
+ } else {
+ b[1] = 0;
+ b[2] = 0;
+ b[3] = 0;
+ }
+ b[0] = 0x80;
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int f, port, len;
+ u8 *b;
+
+ for (f = 0; f < data_blocks; f++) {
+ b = (u8 *)&buffer[0];
+ port = b[3] >> 4;
+ len = b[3] & 0x0f;
+
+ if (port < p->midi_ports && p->midi[port] && len > 0)
+ snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ /* This protocol delivers 24 bit data in 32bit data channel. */
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+
+ return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ if (port < p->midi_ports)
+ ACCESS_ONCE(p->midi[port]) = midi;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks;
+ } else {
+ pcm_frames = 0;
+ }
+
+ read_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks;
+ } else {
+ write_pcm_silence(s, buffer, data_blocks);
+ pcm_frames = 0;
+ }
+
+ write_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+ enum cip_flags flags;
+
+ /* Use different mode between incoming/outgoing. */
+ if (dir == AMDTP_IN_STREAM) {
+ flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK;
+ process_data_blocks = process_tx_data_blocks;
+ } else {
+ flags = CIP_BLOCKING;
+ process_data_blocks = process_rx_data_blocks;
+ }
+
+ return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+ process_data_blocks, sizeof(struct amdtp_dot));
+}
+
+void amdtp_dot_reset(struct amdtp_stream *s)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ p->state.carry = 0x00;
+ p->state.idx = 0x00;
+ p->state.off = 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c
new file mode 100644
index 0000000..f188e47
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-hwdep.c
@@ -0,0 +1,200 @@
+/*
+ * digi00x-hwdep.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ * 4.get asynchronous messaging
+ */
+
+#include "digi00x.h"
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
+ loff_t *offset)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+ DEFINE_WAIT(wait);
+ union snd_firewire_event event;
+
+ spin_lock_irq(&dg00x->lock);
+
+ while (!dg00x->dev_lock_changed && dg00x->msg == 0) {
+ prepare_to_wait(&dg00x->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&dg00x->lock);
+ schedule();
+ finish_wait(&dg00x->hwdep_wait, &wait);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ spin_lock_irq(&dg00x->lock);
+ }
+
+ memset(&event, 0, sizeof(event));
+ if (dg00x->dev_lock_changed) {
+ event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+ event.lock_status.status = (dg00x->dev_lock_count > 0);
+ dg00x->dev_lock_changed = false;
+
+ count = min_t(long, count, sizeof(event.lock_status));
+ } else {
+ event.digi00x_message.type =
+ SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE;
+ event.digi00x_message.message = dg00x->msg;
+ dg00x->msg = 0;
+
+ count = min_t(long, count, sizeof(event.digi00x_message));
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ if (copy_to_user(buf, &event, count))
+ return -EFAULT;
+
+ return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+ poll_table *wait)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+ unsigned int events;
+
+ poll_wait(file, &dg00x->hwdep_wait, wait);
+
+ spin_lock_irq(&dg00x->lock);
+ if (dg00x->dev_lock_changed || dg00x->msg)
+ events = POLLIN | POLLRDNORM;
+ else
+ events = 0;
+ spin_unlock_irq(&dg00x->lock);
+
+ return events;
+}
+
+static int hwdep_get_info(struct snd_dg00x *dg00x, void __user *arg)
+{
+ struct fw_device *dev = fw_parent_device(dg00x->unit);
+ struct snd_firewire_get_info info;
+
+ memset(&info, 0, sizeof(info));
+ info.type = SNDRV_FIREWIRE_TYPE_DIGI00X;
+ info.card = dev->card->index;
+ *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+ *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+ strlcpy(info.device_name, dev_name(&dev->device),
+ sizeof(info.device_name));
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+static int hwdep_lock(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ if (dg00x->dev_lock_count == 0) {
+ dg00x->dev_lock_count = -1;
+ err = 0;
+ } else {
+ err = -EBUSY;
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ return err;
+}
+
+static int hwdep_unlock(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ if (dg00x->dev_lock_count == -1) {
+ dg00x->dev_lock_count = 0;
+ err = 0;
+ } else {
+ err = -EBADFD;
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+
+ spin_lock_irq(&dg00x->lock);
+ if (dg00x->dev_lock_count == -1)
+ dg00x->dev_lock_count = 0;
+ spin_unlock_irq(&dg00x->lock);
+
+ return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+
+ switch (cmd) {
+ case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+ return hwdep_get_info(dg00x, (void __user *)arg);
+ case SNDRV_FIREWIRE_IOCTL_LOCK:
+ return hwdep_lock(dg00x);
+ case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+ return hwdep_unlock(dg00x);
+ default:
+ return -ENOIOCTLCMD;
+ }
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hwdep_ioctl(hwdep, file, cmd,
+ (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+ .read = hwdep_read,
+ .release = hwdep_release,
+ .poll = hwdep_poll,
+ .ioctl = hwdep_ioctl,
+ .ioctl_compat = hwdep_compat_ioctl,
+};
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x)
+{
+ struct snd_hwdep *hwdep;
+ int err;
+
+ err = snd_hwdep_new(dg00x->card, "Digi00x", 0, &hwdep);
+ if (err < 0)
+ return err;
+
+ strcpy(hwdep->name, "Digi00x");
+ hwdep->iface = SNDRV_HWDEP_IFACE_FW_DIGI00X;
+ hwdep->ops = hwdep_ops;
+ hwdep->private_data = dg00x;
+ hwdep->exclusive = true;
+
+ return err;
+}
diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c
new file mode 100644
index 0000000..1a72a38
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-midi.c
@@ -0,0 +1,223 @@
+/*
+ * digi00x-midi.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int midi_phys_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ int err;
+
+ err = snd_dg00x_stream_lock_try(dg00x);
+ if (err < 0)
+ return err;
+
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ err = snd_dg00x_stream_start_duplex(dg00x, 0);
+ mutex_unlock(&dg00x->mutex);
+ if (err < 0)
+ snd_dg00x_stream_lock_release(dg00x);
+
+ return err;
+}
+
+static int midi_phys_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter--;
+ snd_dg00x_stream_stop_duplex(dg00x);
+ mutex_unlock(&dg00x->mutex);
+
+ snd_dg00x_stream_lock_release(dg00x);
+ return 0;
+}
+
+static void midi_phys_capture_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number,
+ substream);
+ else
+ amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number,
+ NULL);
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static void midi_phys_playback_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number,
+ substream);
+ else
+ amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number,
+ NULL);
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_phys_capture_ops = {
+ .open = midi_phys_open,
+ .close = midi_phys_close,
+ .trigger = midi_phys_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_phys_playback_ops = {
+ .open = midi_phys_open,
+ .close = midi_phys_close,
+ .trigger = midi_phys_playback_trigger,
+};
+
+static int midi_ctl_open(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_ctl_capture_close(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_ctl_playback_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+
+ snd_fw_async_midi_port_finish(&dg00x->out_control);
+
+ return 0;
+}
+
+static void midi_ctl_capture_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ dg00x->in_control = substream;
+ else
+ dg00x->in_control = NULL;
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static void midi_ctl_playback_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ snd_fw_async_midi_port_run(&dg00x->out_control, substream);
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_ctl_capture_ops = {
+ .open = midi_ctl_open,
+ .close = midi_ctl_capture_close,
+ .trigger = midi_ctl_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_ctl_playback_ops = {
+ .open = midi_ctl_open,
+ .close = midi_ctl_playback_close,
+ .trigger = midi_ctl_playback_trigger,
+};
+
+static void set_midi_substream_names(struct snd_dg00x *dg00x,
+ struct snd_rawmidi_str *str,
+ bool is_ctl)
+{
+ struct snd_rawmidi_substream *subs;
+
+ list_for_each_entry(subs, &str->substreams, list) {
+ if (!is_ctl)
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ dg00x->card->shortname, subs->number + 1);
+ else
+ /* This port is for asynchronous transaction. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s control",
+ dg00x->card->shortname);
+ }
+}
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x)
+{
+ struct snd_rawmidi *rmidi[2];
+ struct snd_rawmidi_str *str;
+ unsigned int i;
+ int err;
+
+ /* Add physical midi ports. */
+ err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 0,
+ DOT_MIDI_OUT_PORTS, DOT_MIDI_IN_PORTS, &rmidi[0]);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi[0]->name, sizeof(rmidi[0]->name),
+ "%s MIDI", dg00x->card->shortname);
+
+ snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_phys_capture_ops);
+ snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_phys_playback_ops);
+
+ /* Add a pair of control midi ports. */
+ err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 1,
+ 1, 1, &rmidi[1]);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi[1]->name, sizeof(rmidi[1]->name),
+ "%s control", dg00x->card->shortname);
+
+ snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_ctl_capture_ops);
+ snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_ctl_playback_ops);
+
+ for (i = 0; i < ARRAY_SIZE(rmidi); i++) {
+ rmidi[i]->private_data = dg00x;
+
+ rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+ set_midi_substream_names(dg00x, str, i);
+
+ rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+ set_midi_substream_names(dg00x, str, i);
+
+ rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+ }
+
+ return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
new file mode 100644
index 0000000..cac28f7
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -0,0 +1,373 @@
+/*
+ * digi00x-pcm.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ const struct snd_interval *c =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1,
+ };
+ unsigned int i;
+
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (!snd_interval_test(c,
+ snd_dg00x_stream_pcm_channels[i]))
+ continue;
+
+ t.min = min(t.min, snd_dg00x_stream_rates[i]);
+ t.max = max(t.max, snd_dg00x_stream_rates[i]);
+ }
+
+ return snd_interval_refine(r, &t);
+}
+
+static int hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ const struct snd_interval *r =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1,
+ };
+ unsigned int i;
+
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (!snd_interval_test(r, snd_dg00x_stream_rates[i]))
+ continue;
+
+ t.min = min(t.min, snd_dg00x_stream_pcm_channels[i]);
+ t.max = max(t.max, snd_dg00x_stream_pcm_channels[i]);
+ }
+
+ return snd_interval_refine(c, &t);
+}
+
+static int pcm_init_hw_params(struct snd_dg00x *dg00x,
+ struct snd_pcm_substream *substream)
+{
+ static const struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 18,
+ .period_bytes_min = 4 * 18,
+ .period_bytes_max = 4 * 18 * 2048,
+ .buffer_bytes_max = 4 * 18 * 2048 * 2,
+ .periods_min = 2,
+ .periods_max = UINT_MAX,
+ };
+ struct amdtp_stream *s;
+ int err;
+
+ substream->runtime->hw = hardware;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+ s = &dg00x->tx_stream;
+ } else {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16 |
+ SNDRV_PCM_FMTBIT_S32;
+ s = &dg00x->rx_stream;
+ }
+
+ err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_channels, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_rate, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+
+ return amdtp_dot_add_pcm_hw_constraints(s, substream->runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ enum snd_dg00x_clock clock;
+ bool detect;
+ unsigned int rate;
+ int err;
+
+ err = snd_dg00x_stream_lock_try(dg00x);
+ if (err < 0)
+ goto end;
+
+ err = pcm_init_hw_params(dg00x, substream);
+ if (err < 0)
+ goto err_locked;
+
+ /* Check current clock source. */
+ err = snd_dg00x_stream_get_clock(dg00x, &clock);
+ if (err < 0)
+ goto err_locked;
+ if (clock != SND_DG00X_CLOCK_INTERNAL) {
+ err = snd_dg00x_stream_check_external_clock(dg00x, &detect);
+ if (err < 0)
+ goto err_locked;
+ if (!detect) {
+ err = -EBUSY;
+ goto err_locked;
+ }
+ }
+
+ if ((clock != SND_DG00X_CLOCK_INTERNAL) ||
+ amdtp_stream_pcm_running(&dg00x->rx_stream) ||
+ amdtp_stream_pcm_running(&dg00x->tx_stream)) {
+ err = snd_dg00x_stream_get_external_rate(dg00x, &rate);
+ if (err < 0)
+ goto err_locked;
+ substream->runtime->hw.rate_min = rate;
+ substream->runtime->hw.rate_max = rate;
+ }
+
+ snd_pcm_set_sync(substream);
+end:
+ return err;
+err_locked:
+ snd_dg00x_stream_lock_release(dg00x);
+ return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ snd_dg00x_stream_lock_release(dg00x);
+
+ return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ mutex_unlock(&dg00x->mutex);
+ }
+
+ amdtp_dot_set_pcm_format(&dg00x->tx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ mutex_unlock(&dg00x->mutex);
+ }
+
+ amdtp_dot_set_pcm_format(&dg00x->rx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ mutex_lock(&dg00x->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ dg00x->substreams_counter--;
+
+ snd_dg00x_stream_stop_duplex(dg00x);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ mutex_lock(&dg00x->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ dg00x->substreams_counter--;
+
+ snd_dg00x_stream_stop_duplex(dg00x);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&dg00x->mutex);
+
+ err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&dg00x->tx_stream);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&dg00x->mutex);
+
+ err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+ if (err >= 0) {
+ amdtp_stream_pcm_prepare(&dg00x->rx_stream);
+ amdtp_dot_reset(&dg00x->rx_stream);
+ }
+
+ mutex_unlock(&dg00x->mutex);
+
+ return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&dg00x->tx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&dg00x->tx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&dg00x->rx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&dg00x->rx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_dg00x *dg00x = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&dg00x->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_dg00x *dg00x = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&dg00x->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_capture_hw_params,
+ .hw_free = pcm_capture_hw_free,
+ .prepare = pcm_capture_prepare,
+ .trigger = pcm_capture_trigger,
+ .pointer = pcm_capture_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_playback_hw_params,
+ .hw_free = pcm_playback_hw_free,
+ .prepare = pcm_playback_prepare,
+ .trigger = pcm_playback_trigger,
+ .pointer = pcm_playback_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(dg00x->card, dg00x->card->driver, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = dg00x;
+ snprintf(pcm->name, sizeof(pcm->name),
+ "%s PCM", dg00x->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+ return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c
new file mode 100644
index 0000000..a1d601f
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-proc.c
@@ -0,0 +1,99 @@
+/*
+ * digi00x-proc.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int get_optical_iface_mode(struct snd_dg00x *dg00x,
+ enum snd_dg00x_optical_mode *mode)
+{
+ __be32 data;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_OPT_IFACE_MODE,
+ &data, sizeof(data), 0);
+ if (err >= 0)
+ *mode = be32_to_cpu(data) & 0x01;
+
+ return err;
+}
+
+static void proc_read_clock(struct snd_info_entry *entry,
+ struct snd_info_buffer *buf)
+{
+ static const char *const source_name[] = {
+ [SND_DG00X_CLOCK_INTERNAL] = "internal",
+ [SND_DG00X_CLOCK_SPDIF] = "s/pdif",
+ [SND_DG00X_CLOCK_ADAT] = "adat",
+ [SND_DG00X_CLOCK_WORD] = "word clock",
+ };
+ static const char *const optical_name[] = {
+ [SND_DG00X_OPT_IFACE_MODE_ADAT] = "adat",
+ [SND_DG00X_OPT_IFACE_MODE_SPDIF] = "s/pdif",
+ };
+ struct snd_dg00x *dg00x = entry->private_data;
+ enum snd_dg00x_optical_mode mode;
+ unsigned int rate;
+ enum snd_dg00x_clock clock;
+ bool detect;
+
+ if (get_optical_iface_mode(dg00x, &mode) < 0)
+ return;
+ if (snd_dg00x_stream_get_local_rate(dg00x, &rate) < 0)
+ return;
+ if (snd_dg00x_stream_get_clock(dg00x, &clock) < 0)
+ return;
+
+ snd_iprintf(buf, "Optical mode: %s\n", optical_name[mode]);
+ snd_iprintf(buf, "Sampling Rate: %d\n", rate);
+ snd_iprintf(buf, "Clock Source: %s\n", source_name[clock]);
+
+ if (clock == SND_DG00X_CLOCK_INTERNAL)
+ return;
+
+ if (snd_dg00x_stream_check_external_clock(dg00x, &detect) < 0)
+ return;
+ snd_iprintf(buf, "External source: %s\n", detect ? "detected" : "not");
+ if (!detect)
+ return;
+
+ if (snd_dg00x_stream_get_external_rate(dg00x, &rate) >= 0)
+ snd_iprintf(buf, "External sampling rate: %d\n", rate);
+}
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x)
+{
+ struct snd_info_entry *root, *entry;
+
+ /*
+ * All nodes are automatically removed at snd_card_disconnect(),
+ * by following to link list.
+ */
+ root = snd_info_create_card_entry(dg00x->card, "firewire",
+ dg00x->card->proc_root);
+ if (root == NULL)
+ return;
+
+ root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(root) < 0) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ entry = snd_info_create_card_entry(dg00x->card, "clock", root);
+ if (entry == NULL) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ snd_info_set_text_ops(entry, dg00x, proc_read_clock);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ snd_info_free_entry(root);
+ }
+}
diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c
new file mode 100644
index 0000000..4d3b4eb
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-stream.c
@@ -0,0 +1,422 @@
+/*
+ * digi00x-stream.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+#define CALLBACK_TIMEOUT 500
+
+const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT] = {
+ [SND_DG00X_RATE_44100] = 44100,
+ [SND_DG00X_RATE_48000] = 48000,
+ [SND_DG00X_RATE_88200] = 88200,
+ [SND_DG00X_RATE_96000] = 96000,
+};
+
+/* Multi Bit Linear Audio data channels for each sampling transfer frequency. */
+const unsigned int
+snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT] = {
+ /* Analog/ADAT/SPDIF */
+ [SND_DG00X_RATE_44100] = (8 + 8 + 2),
+ [SND_DG00X_RATE_48000] = (8 + 8 + 2),
+ /* Analog/SPDIF */
+ [SND_DG00X_RATE_88200] = (8 + 2),
+ [SND_DG00X_RATE_96000] = (8 + 2),
+};
+
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x, unsigned int *rate)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = be32_to_cpu(reg) & 0x0f;
+ if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+ *rate = snd_dg00x_stream_rates[data];
+ else
+ err = -EIO;
+
+ return err;
+}
+
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ __be32 reg;
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_dg00x_stream_rates); i++) {
+ if (rate == snd_dg00x_stream_rates[i])
+ break;
+ }
+ if (i == ARRAY_SIZE(snd_dg00x_stream_rates))
+ return -EINVAL;
+
+ reg = cpu_to_be32(i);
+ return snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+ ®, sizeof(reg), 0);
+}
+
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+ enum snd_dg00x_clock *clock)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_CLOCK_SOURCE,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ *clock = be32_to_cpu(reg) & 0x0f;
+ if (*clock >= SND_DG00X_CLOCK_COUNT)
+ err = -EIO;
+
+ return err;
+}
+
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, bool *detect)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_DETECT_EXTERNAL,
+ ®, sizeof(reg), 0);
+ if (err >= 0)
+ *detect = be32_to_cpu(reg) > 0;
+
+ return err;
+}
+
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_EXTERNAL_RATE,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = be32_to_cpu(reg) & 0x0f;
+ if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+ *rate = snd_dg00x_stream_rates[data];
+ /* This means desync. */
+ else
+ err = -EBUSY;
+
+ return err;
+}
+
+static void finish_session(struct snd_dg00x *dg00x)
+{
+ __be32 data = cpu_to_be32(0x00000003);
+
+ snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET,
+ &data, sizeof(data), 0);
+}
+
+static int begin_session(struct snd_dg00x *dg00x)
+{
+ __be32 data;
+ u32 curr;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_STATE,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+ curr = be32_to_cpu(data);
+
+ if (curr == 0)
+ curr = 2;
+
+ curr--;
+ while (curr > 0) {
+ data = cpu_to_be32(curr);
+ err = snd_fw_transaction(dg00x->unit,
+ TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE +
+ DG00X_OFFSET_STREAMING_SET,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+
+ msleep(20);
+ curr--;
+ }
+
+ return 0;
+error:
+ finish_session(dg00x);
+ return err;
+}
+
+static void release_resources(struct snd_dg00x *dg00x)
+{
+ __be32 data = 0;
+
+ /* Unregister isochronous channels for both direction. */
+ snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+ &data, sizeof(data), 0);
+
+ /* Release isochronous resources. */
+ fw_iso_resources_free(&dg00x->tx_resources);
+ fw_iso_resources_free(&dg00x->rx_resources);
+}
+
+static int keep_resources(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ unsigned int i;
+ __be32 data;
+ int err;
+
+ /* Check sampling rate. */
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (snd_dg00x_stream_rates[i] == rate)
+ break;
+ }
+ if (i == SND_DG00X_RATE_COUNT)
+ return -EINVAL;
+
+ /* Keep resources for out-stream. */
+ err = amdtp_dot_set_parameters(&dg00x->rx_stream, rate,
+ snd_dg00x_stream_pcm_channels[i]);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&dg00x->rx_resources,
+ amdtp_stream_get_max_payload(&dg00x->rx_stream),
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ return err;
+
+ /* Keep resources for in-stream. */
+ err = amdtp_dot_set_parameters(&dg00x->tx_stream, rate,
+ snd_dg00x_stream_pcm_channels[i]);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&dg00x->tx_resources,
+ amdtp_stream_get_max_payload(&dg00x->tx_stream),
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ /* Register isochronous channels for both direction. */
+ data = cpu_to_be32((dg00x->tx_resources.channel << 16) |
+ dg00x->rx_resources.channel);
+ err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ release_resources(dg00x);
+ return err;
+}
+
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ /* For out-stream. */
+ err = fw_iso_resources_init(&dg00x->rx_resources, dg00x->unit);
+ if (err < 0)
+ goto error;
+ err = amdtp_dot_init(&dg00x->rx_stream, dg00x->unit, AMDTP_OUT_STREAM);
+ if (err < 0)
+ goto error;
+
+ /* For in-stream. */
+ err = fw_iso_resources_init(&dg00x->tx_resources, dg00x->unit);
+ if (err < 0)
+ goto error;
+ err = amdtp_dot_init(&dg00x->tx_stream, dg00x->unit, AMDTP_IN_STREAM);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ snd_dg00x_stream_destroy_duplex(dg00x);
+ return err;
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x)
+{
+ amdtp_stream_destroy(&dg00x->rx_stream);
+ fw_iso_resources_destroy(&dg00x->rx_resources);
+
+ amdtp_stream_destroy(&dg00x->tx_stream);
+ fw_iso_resources_destroy(&dg00x->tx_resources);
+}
+
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ unsigned int curr_rate;
+ int err = 0;
+
+ if (dg00x->substreams_counter == 0)
+ goto end;
+
+ /* Check current sampling rate. */
+ err = snd_dg00x_stream_get_local_rate(dg00x, &curr_rate);
+ if (err < 0)
+ goto error;
+ if (rate == 0)
+ rate = curr_rate;
+ if (curr_rate != rate ||
+ amdtp_streaming_error(&dg00x->tx_stream) ||
+ amdtp_streaming_error(&dg00x->rx_stream)) {
+ finish_session(dg00x);
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ release_resources(dg00x);
+ }
+
+ /*
+ * No packets are transmitted without receiving packets, reagardless of
+ * which source of clock is used.
+ */
+ if (!amdtp_stream_running(&dg00x->rx_stream)) {
+ err = snd_dg00x_stream_set_local_rate(dg00x, rate);
+ if (err < 0)
+ goto error;
+
+ err = keep_resources(dg00x, rate);
+ if (err < 0)
+ goto error;
+
+ err = begin_session(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = amdtp_stream_start(&dg00x->rx_stream,
+ dg00x->rx_resources.channel,
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&dg00x->rx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ /*
+ * The value of SYT field in transmitted packets is always 0x0000. Thus,
+ * duplex streams with timestamp synchronization cannot be built.
+ */
+ if (!amdtp_stream_running(&dg00x->tx_stream)) {
+ err = amdtp_stream_start(&dg00x->tx_stream,
+ dg00x->tx_resources.channel,
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&dg00x->tx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+end:
+ return err;
+error:
+ finish_session(dg00x);
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ release_resources(dg00x);
+
+ return err;
+}
+
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x)
+{
+ if (dg00x->substreams_counter > 0)
+ return;
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ finish_session(dg00x);
+ release_resources(dg00x);
+
+ /*
+ * Just after finishing the session, the device may lost transmitting
+ * functionality for a short time.
+ */
+ msleep(50);
+}
+
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x)
+{
+ fw_iso_resources_update(&dg00x->tx_resources);
+ fw_iso_resources_update(&dg00x->rx_resources);
+
+ amdtp_stream_update(&dg00x->tx_stream);
+ amdtp_stream_update(&dg00x->rx_stream);
+}
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x)
+{
+ dg00x->dev_lock_changed = true;
+ wake_up(&dg00x->hwdep_wait);
+}
+
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ /* user land lock this */
+ if (dg00x->dev_lock_count < 0) {
+ err = -EBUSY;
+ goto end;
+ }
+
+ /* this is the first time */
+ if (dg00x->dev_lock_count++ == 0)
+ snd_dg00x_stream_lock_changed(dg00x);
+ err = 0;
+end:
+ spin_unlock_irq(&dg00x->lock);
+ return err;
+}
+
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x)
+{
+ spin_lock_irq(&dg00x->lock);
+
+ if (WARN_ON(dg00x->dev_lock_count <= 0))
+ goto end;
+ if (--dg00x->dev_lock_count == 0)
+ snd_dg00x_stream_lock_changed(dg00x);
+end:
+ spin_unlock_irq(&dg00x->lock);
+}
diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c
new file mode 100644
index 0000000..554324d
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-transaction.c
@@ -0,0 +1,137 @@
+/*
+ * digi00x-transaction.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/asound.h>
+#include "digi00x.h"
+
+static int fill_midi_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+ int bytes;
+
+ buf[0] = 0x80;
+ bytes = snd_rawmidi_transmit_peek(substream, buf + 1, 2);
+ if (bytes >= 0)
+ buf[3] = 0xc0 | bytes;
+
+ return bytes;
+}
+
+static void handle_midi_control(struct snd_dg00x *dg00x, __be32 *buf,
+ unsigned int length)
+{
+ struct snd_rawmidi_substream *substream;
+ unsigned int i;
+ unsigned int len;
+ u8 *b;
+
+ substream = ACCESS_ONCE(dg00x->in_control);
+ if (substream == NULL)
+ return;
+
+ length /= 4;
+
+ for (i = 0; i < length; i++) {
+ b = (u8 *)&buf[i];
+ len = b[3] & 0xf;
+ if (len > 0)
+ snd_rawmidi_receive(dg00x->in_control, b + 1, len);
+ }
+}
+
+static void handle_unknown_message(struct snd_dg00x *dg00x,
+ unsigned long long offset, __be32 *buf)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+ dg00x->msg = be32_to_cpu(*buf);
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+
+ wake_up(&dg00x->hwdep_wait);
+}
+
+static void handle_message(struct fw_card *card, struct fw_request *request,
+ int tcode, int destination, int source,
+ int generation, unsigned long long offset,
+ void *data, size_t length, void *callback_data)
+{
+ struct snd_dg00x *dg00x = callback_data;
+ __be32 *buf = (__be32 *)data;
+
+ if (offset == dg00x->async_handler.offset)
+ handle_unknown_message(dg00x, offset, buf);
+ else if (offset == dg00x->async_handler.offset + 4)
+ handle_midi_control(dg00x, buf, length);
+
+ fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x)
+{
+ struct fw_device *device = fw_parent_device(dg00x->unit);
+ __be32 data[2];
+ int err;
+
+ /* Unknown. 4bytes. */
+ data[0] = cpu_to_be32((device->card->node_id << 16) |
+ (dg00x->async_handler.offset >> 32));
+ data[1] = cpu_to_be32(dg00x->async_handler.offset);
+ err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MESSAGE_ADDR,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return err;
+
+ /* Asynchronous transactions for MIDI control message. */
+ data[0] = cpu_to_be32((device->card->node_id << 16) |
+ (dg00x->async_handler.offset >> 32));
+ data[1] = cpu_to_be32(dg00x->async_handler.offset + 4);
+ return snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MIDI_CTL_ADDR,
+ &data, sizeof(data), 0);
+}
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x)
+{
+ static const struct fw_address_region resp_register_region = {
+ .start = 0xffffe0000000ull,
+ .end = 0xffffe000ffffull,
+ };
+ int err;
+
+ dg00x->async_handler.length = 12;
+ dg00x->async_handler.address_callback = handle_message;
+ dg00x->async_handler.callback_data = dg00x;
+
+ err = fw_core_add_address_handler(&dg00x->async_handler,
+ &resp_register_region);
+ if (err < 0)
+ return err;
+
+ err = snd_dg00x_transaction_reregister(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_fw_async_midi_port_init(&dg00x->out_control, dg00x->unit,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MMC,
+ 4, fill_midi_message);
+ if (err < 0)
+ goto error;
+
+ return err;
+error:
+ fw_core_remove_address_handler(&dg00x->async_handler);
+ dg00x->async_handler.address_callback = NULL;
+ return err;
+}
+
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x)
+{
+ snd_fw_async_midi_port_destroy(&dg00x->out_control);
+ fw_core_remove_address_handler(&dg00x->async_handler);
+}
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
new file mode 100644
index 0000000..1f33b7a
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.c
@@ -0,0 +1,170 @@
+/*
+ * digi00x.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+MODULE_DESCRIPTION("Digidesign Digi 002/003 family Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+#define VENDOR_DIGIDESIGN 0x00a07e
+#define MODEL_DIGI00X 0x000002
+
+static int name_card(struct snd_dg00x *dg00x)
+{
+ struct fw_device *fw_dev = fw_parent_device(dg00x->unit);
+ char name[32] = {0};
+ char *model;
+ int err;
+
+ err = fw_csr_string(dg00x->unit->directory, CSR_MODEL, name,
+ sizeof(name));
+ if (err < 0)
+ return err;
+
+ model = skip_spaces(name);
+
+ strcpy(dg00x->card->driver, "Digi00x");
+ strcpy(dg00x->card->shortname, model);
+ strcpy(dg00x->card->mixername, model);
+ snprintf(dg00x->card->longname, sizeof(dg00x->card->longname),
+ "Digidesign %s, GUID %08x%08x at %s, S%d", model,
+ fw_dev->config_rom[3], fw_dev->config_rom[4],
+ dev_name(&dg00x->unit->device), 100 << fw_dev->max_speed);
+
+ return 0;
+}
+
+static void dg00x_card_free(struct snd_card *card)
+{
+ struct snd_dg00x *dg00x = card->private_data;
+
+ snd_dg00x_stream_destroy_duplex(dg00x);
+ snd_dg00x_transaction_unregister(dg00x);
+
+ fw_unit_put(dg00x->unit);
+
+ mutex_destroy(&dg00x->mutex);
+}
+
+static int snd_dg00x_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_card *card;
+ struct snd_dg00x *dg00x;
+ int err;
+
+ /* create card */
+ err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+ sizeof(struct snd_dg00x), &card);
+ if (err < 0)
+ return err;
+ card->private_free = dg00x_card_free;
+
+ /* initialize myself */
+ dg00x = card->private_data;
+ dg00x->card = card;
+ dg00x->unit = fw_unit_get(unit);
+
+ mutex_init(&dg00x->mutex);
+ spin_lock_init(&dg00x->lock);
+ init_waitqueue_head(&dg00x->hwdep_wait);
+
+ err = name_card(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_stream_init_duplex(dg00x);
+ if (err < 0)
+ goto error;
+
+ snd_dg00x_proc_init(dg00x);
+
+ err = snd_dg00x_create_pcm_devices(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_create_midi_devices(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_create_hwdep_device(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_transaction_register(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ dev_set_drvdata(&unit->device, dg00x);
+
+ return err;
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static void snd_dg00x_update(struct fw_unit *unit)
+{
+ struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+ snd_dg00x_transaction_reregister(dg00x);
+
+ mutex_lock(&dg00x->mutex);
+ snd_dg00x_stream_update_duplex(dg00x);
+ mutex_unlock(&dg00x->mutex);
+}
+
+static void snd_dg00x_remove(struct fw_unit *unit)
+{
+ struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(dg00x->card);
+}
+
+static const struct ieee1394_device_id snd_dg00x_id_table[] = {
+ /* Both of 002/003 use the same ID. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = VENDOR_DIGIDESIGN,
+ .model_id = MODEL_DIGI00X,
+ },
+ {}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table);
+
+static struct fw_driver dg00x_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "snd-firewire-digi00x",
+ .bus = &fw_bus_type,
+ },
+ .probe = snd_dg00x_probe,
+ .update = snd_dg00x_update,
+ .remove = snd_dg00x_remove,
+ .id_table = snd_dg00x_id_table,
+};
+
+static int __init snd_dg00x_init(void)
+{
+ return driver_register(&dg00x_driver.driver);
+}
+
+static void __exit snd_dg00x_exit(void)
+{
+ driver_unregister(&dg00x_driver.driver);
+}
+
+module_init(snd_dg00x_init);
+module_exit(snd_dg00x_exit);
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
new file mode 100644
index 0000000..907e739
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.h
@@ -0,0 +1,157 @@
+/*
+ * digi00x.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_DIGI00X_H_INCLUDED
+#define SOUND_DIGI00X_H_INCLUDED
+
+#include <linux/compat.h>
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../iso-resources.h"
+#include "../amdtp-stream.h"
+
+struct snd_dg00x {
+ struct snd_card *card;
+ struct fw_unit *unit;
+
+ struct mutex mutex;
+ spinlock_t lock;
+
+ struct amdtp_stream tx_stream;
+ struct fw_iso_resources tx_resources;
+
+ struct amdtp_stream rx_stream;
+ struct fw_iso_resources rx_resources;
+
+ unsigned int substreams_counter;
+
+ /* for uapi */
+ int dev_lock_count;
+ bool dev_lock_changed;
+ wait_queue_head_t hwdep_wait;
+
+ /* For asynchronous messages. */
+ struct fw_address_handler async_handler;
+ u32 msg;
+
+ /* For asynchronous MIDI controls. */
+ struct snd_rawmidi_substream *in_control;
+ struct snd_fw_async_midi_port out_control;
+};
+
+#define DG00X_ADDR_BASE 0xffffe0000000ull
+
+#define DG00X_OFFSET_STREAMING_STATE 0x0000
+#define DG00X_OFFSET_STREAMING_SET 0x0004
+#define DG00X_OFFSET_MIDI_CTL_ADDR 0x0008
+/* For LSB of the address 0x000c */
+/* unknown 0x0010 */
+#define DG00X_OFFSET_MESSAGE_ADDR 0x0014
+/* For LSB of the address 0x0018 */
+/* unknown 0x001c */
+/* unknown 0x0020 */
+/* not used 0x0024--0x00ff */
+#define DG00X_OFFSET_ISOC_CHANNELS 0x0100
+/* unknown 0x0104 */
+/* unknown 0x0108 */
+/* unknown 0x010c */
+#define DG00X_OFFSET_LOCAL_RATE 0x0110
+#define DG00X_OFFSET_EXTERNAL_RATE 0x0114
+#define DG00X_OFFSET_CLOCK_SOURCE 0x0118
+#define DG00X_OFFSET_OPT_IFACE_MODE 0x011c
+/* unknown 0x0120 */
+/* Mixer control on/off 0x0124 */
+/* unknown 0x0128 */
+#define DG00X_OFFSET_DETECT_EXTERNAL 0x012c
+/* unknown 0x0138 */
+#define DG00X_OFFSET_MMC 0x0400
+
+enum snd_dg00x_rate {
+ SND_DG00X_RATE_44100 = 0,
+ SND_DG00X_RATE_48000,
+ SND_DG00X_RATE_88200,
+ SND_DG00X_RATE_96000,
+ SND_DG00X_RATE_COUNT,
+};
+
+enum snd_dg00x_clock {
+ SND_DG00X_CLOCK_INTERNAL = 0,
+ SND_DG00X_CLOCK_SPDIF,
+ SND_DG00X_CLOCK_ADAT,
+ SND_DG00X_CLOCK_WORD,
+ SND_DG00X_CLOCK_COUNT,
+};
+
+enum snd_dg00x_optical_mode {
+ SND_DG00X_OPT_IFACE_MODE_ADAT = 0,
+ SND_DG00X_OPT_IFACE_MODE_SPDIF,
+ SND_DG00X_OPT_IFACE_MODE_COUNT,
+};
+
+#define DOT_MIDI_IN_PORTS 1
+#define DOT_MIDI_OUT_PORTS 2
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir);
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels);
+void amdtp_dot_reset(struct amdtp_stream *s);
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi);
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x);
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x);
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x);
+
+extern const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT];
+extern const unsigned int snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT];
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate);
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate);
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate);
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+ enum snd_dg00x_clock *clock);
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x,
+ bool *detect);
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate);
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x);
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x);
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x);
+#endif
diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c
index 0619597..cce1976 100644
--- a/sound/firewire/fcp.c
+++ b/sound/firewire/fcp.c
@@ -17,7 +17,7 @@
#include <linux/delay.h>
#include "fcp.h"
#include "lib.h"
-#include "amdtp.h"
+#include "amdtp-stream.h"
#define CTS_AVC 0x00
diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile
index 0c74408..15ef7f7 100644
--- a/sound/firewire/fireworks/Makefile
+++ b/sound/firewire/fireworks/Makefile
@@ -1,4 +1,4 @@
snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \
fireworks_stream.o fireworks_proc.o fireworks_midi.o \
fireworks_pcm.o fireworks_hwdep.o fireworks.o
-obj-m += snd-fireworks.o
+obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index c94a432..d5b19bc 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -138,12 +138,12 @@
efw->midi_out_ports = hwinfo->midi_out_ports;
efw->midi_in_ports = hwinfo->midi_in_ports;
- if (hwinfo->amdtp_tx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_tx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_tx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM) {
+ if (hwinfo->amdtp_tx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_tx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_tx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 084d414..c7cb7de 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -29,7 +29,7 @@
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
#include "../lib.h"
diff --git a/sound/firewire/fireworks/fireworks_command.c b/sound/firewire/fireworks/fireworks_command.c
index 166f805..94bab04 100644
--- a/sound/firewire/fireworks/fireworks_command.c
+++ b/sound/firewire/fireworks/fireworks_command.c
@@ -257,7 +257,7 @@
struct snd_efw_phys_meters *meters,
unsigned int len)
{
- __be32 *buf = (__be32 *)meters;
+ u32 *buf = (u32 *)meters;
unsigned int i;
int err;
diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c
index cf9c652..fba01bb 100644
--- a/sound/firewire/fireworks/fireworks_midi.c
+++ b/sound/firewire/fireworks/fireworks_midi.c
@@ -73,10 +73,10 @@
spin_lock_irqsave(&efw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&efw->tx_stream,
+ amdtp_am824_midi_trigger(&efw->tx_stream,
substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&efw->tx_stream,
+ amdtp_am824_midi_trigger(&efw->tx_stream,
substrm->number, NULL);
spin_unlock_irqrestore(&efw->lock, flags);
@@ -90,11 +90,11 @@
spin_lock_irqsave(&efw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&efw->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&efw->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&efw->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&efw->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&efw->lock, flags);
}
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index c30b2ff..d27135b 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -159,11 +159,11 @@
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
s = &efw->tx_stream;
pcm_channels = efw->pcm_capture_channels;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
s = &efw->rx_stream;
pcm_channels = efw->pcm_playback_channels;
}
@@ -187,7 +187,7 @@
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
end:
return err;
}
@@ -253,7 +253,8 @@
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&efw->capture_substreams);
- amdtp_stream_set_pcm_format(&efw->tx_stream, params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&efw->tx_stream, params_format(hw_params));
return 0;
}
@@ -270,7 +271,8 @@
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&efw->playback_substreams);
- amdtp_stream_set_pcm_format(&efw->rx_stream, params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&efw->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 7e353f1..759f6e3 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -31,7 +31,7 @@
if (err < 0)
goto end;
- err = amdtp_stream_init(stream, efw->unit, s_dir, CIP_BLOCKING);
+ err = amdtp_am824_init(stream, efw->unit, s_dir, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
cmp_connection_destroy(conn);
@@ -73,8 +73,10 @@
midi_ports = efw->midi_in_ports;
}
- amdtp_stream_set_parameters(stream, sampling_rate,
- pcm_channels, midi_ports);
+ err = amdtp_am824_set_parameters(stream, sampling_rate,
+ pcm_channels, midi_ports, false);
+ if (err < 0)
+ goto end;
/* establish connection via CMP */
err = cmp_connection_establish(conn,
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
index 7409edb..f80aafa 100644
--- a/sound/firewire/lib.c
+++ b/sound/firewire/lib.c
@@ -9,6 +9,7 @@
#include <linux/device.h>
#include <linux/firewire.h>
#include <linux/module.h>
+#include <linux/slab.h>
#include "lib.h"
#define ERROR_RETRY_DELAY_MS 20
@@ -66,6 +67,147 @@
}
EXPORT_SYMBOL(snd_fw_transaction);
+static void async_midi_port_callback(struct fw_card *card, int rcode,
+ void *data, size_t length,
+ void *callback_data)
+{
+ struct snd_fw_async_midi_port *port = callback_data;
+ struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+
+ /* This port is closed. */
+ if (substream == NULL)
+ return;
+
+ if (rcode == RCODE_COMPLETE)
+ snd_rawmidi_transmit_ack(substream, port->consume_bytes);
+ else if (!rcode_is_permanent_error(rcode))
+ /* To start next transaction immediately for recovery. */
+ port->next_ktime = ktime_set(0, 0);
+ else
+ /* Don't continue processing. */
+ port->error = true;
+
+ port->idling = true;
+
+ if (!snd_rawmidi_transmit_empty(substream))
+ schedule_work(&port->work);
+}
+
+static void midi_port_work(struct work_struct *work)
+{
+ struct snd_fw_async_midi_port *port =
+ container_of(work, struct snd_fw_async_midi_port, work);
+ struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+ int generation;
+ int type;
+
+ /* Under transacting or error state. */
+ if (!port->idling || port->error)
+ return;
+
+ /* Nothing to do. */
+ if (substream == NULL || snd_rawmidi_transmit_empty(substream))
+ return;
+
+ /* Do it in next chance. */
+ if (ktime_after(port->next_ktime, ktime_get())) {
+ schedule_work(&port->work);
+ return;
+ }
+
+ /*
+ * Fill the buffer. The callee must use snd_rawmidi_transmit_peek().
+ * Later, snd_rawmidi_transmit_ack() is called.
+ */
+ memset(port->buf, 0, port->len);
+ port->consume_bytes = port->fill(substream, port->buf);
+ if (port->consume_bytes <= 0) {
+ /* Do it in next chance, immediately. */
+ if (port->consume_bytes == 0) {
+ port->next_ktime = ktime_set(0, 0);
+ schedule_work(&port->work);
+ } else {
+ /* Fatal error. */
+ port->error = true;
+ }
+ return;
+ }
+
+ /* Calculate type of transaction. */
+ if (port->len == 4)
+ type = TCODE_WRITE_QUADLET_REQUEST;
+ else
+ type = TCODE_WRITE_BLOCK_REQUEST;
+
+ /* Set interval to next transaction. */
+ port->next_ktime = ktime_add_ns(ktime_get(),
+ port->consume_bytes * 8 * NSEC_PER_SEC / 31250);
+
+ /* Start this transaction. */
+ port->idling = false;
+
+ /*
+ * In Linux FireWire core, when generation is updated with memory
+ * barrier, node id has already been updated. In this module, After
+ * this smp_rmb(), load/store instructions to memory are completed.
+ * Thus, both of generation and node id are available with recent
+ * values. This is a light-serialization solution to handle bus reset
+ * events on IEEE 1394 bus.
+ */
+ generation = port->parent->generation;
+ smp_rmb();
+
+ fw_send_request(port->parent->card, &port->transaction, type,
+ port->parent->node_id, generation,
+ port->parent->max_speed, port->addr,
+ port->buf, port->len, async_midi_port_callback,
+ port);
+}
+
+/**
+ * snd_fw_async_midi_port_init - initialize asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port to initialize
+ * @unit: the target of the asynchronous transaction
+ * @addr: the address to which transactions are transferred
+ * @len: the length of transaction
+ * @fill: the callback function to fill given buffer, and returns the
+ * number of consumed bytes for MIDI message.
+ *
+ */
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+ struct fw_unit *unit, u64 addr, unsigned int len,
+ snd_fw_async_midi_port_fill fill)
+{
+ port->len = DIV_ROUND_UP(len, 4) * 4;
+ port->buf = kzalloc(port->len, GFP_KERNEL);
+ if (port->buf == NULL)
+ return -ENOMEM;
+
+ port->parent = fw_parent_device(unit);
+ port->addr = addr;
+ port->fill = fill;
+ port->idling = true;
+ port->next_ktime = ktime_set(0, 0);
+ port->error = false;
+
+ INIT_WORK(&port->work, midi_port_work);
+
+ return 0;
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_init);
+
+/**
+ * snd_fw_async_midi_port_destroy - free asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port structure
+ */
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port)
+{
+ snd_fw_async_midi_port_finish(port);
+ cancel_work_sync(&port->work);
+ kfree(port->buf);
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_destroy);
+
MODULE_DESCRIPTION("FireWire audio helper functions");
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
MODULE_LICENSE("GPL v2");
diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h
index 02cfabc..f3f6f84 100644
--- a/sound/firewire/lib.h
+++ b/sound/firewire/lib.h
@@ -3,6 +3,8 @@
#include <linux/firewire-constants.h>
#include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/rawmidi.h>
struct fw_unit;
@@ -20,4 +22,58 @@
return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR;
}
+struct snd_fw_async_midi_port;
+typedef int (*snd_fw_async_midi_port_fill)(
+ struct snd_rawmidi_substream *substream,
+ u8 *buf);
+
+struct snd_fw_async_midi_port {
+ struct fw_device *parent;
+ struct work_struct work;
+ bool idling;
+ ktime_t next_ktime;
+ bool error;
+
+ u64 addr;
+ struct fw_transaction transaction;
+
+ u8 *buf;
+ unsigned int len;
+
+ struct snd_rawmidi_substream *substream;
+ snd_fw_async_midi_port_fill fill;
+ unsigned int consume_bytes;
+};
+
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+ struct fw_unit *unit, u64 addr, unsigned int len,
+ snd_fw_async_midi_port_fill fill);
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port);
+
+/**
+ * snd_fw_async_midi_port_run - run transactions for the async MIDI port
+ * @port: the asynchronous MIDI port
+ * @substream: the MIDI substream
+ */
+static inline void
+snd_fw_async_midi_port_run(struct snd_fw_async_midi_port *port,
+ struct snd_rawmidi_substream *substream)
+{
+ if (!port->error) {
+ port->substream = substream;
+ schedule_work(&port->work);
+ }
+}
+
+/**
+ * snd_fw_async_midi_port_finish - finish the asynchronous MIDI port
+ * @port: the asynchronous MIDI port
+ */
+static inline void
+snd_fw_async_midi_port_finish(struct snd_fw_async_midi_port *port)
+{
+ port->substream = NULL;
+ port->error = false;
+}
+
#endif
diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile
index a926850..06ff50f 100644
--- a/sound/firewire/oxfw/Makefile
+++ b/sound/firewire/oxfw/Makefile
@@ -1,3 +1,3 @@
snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \
oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o
-obj-m += snd-oxfw.o
+obj-$(CONFIG_SND_OXFW) += snd-oxfw.o
diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c
index 540a303..8665e10 100644
--- a/sound/firewire/oxfw/oxfw-midi.c
+++ b/sound/firewire/oxfw/oxfw-midi.c
@@ -90,11 +90,11 @@
spin_lock_irqsave(&oxfw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&oxfw->tx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&oxfw->tx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&oxfw->tx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&oxfw->tx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&oxfw->lock, flags);
}
@@ -107,11 +107,11 @@
spin_lock_irqsave(&oxfw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&oxfw->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&oxfw->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&oxfw->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&oxfw->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&oxfw->lock, flags);
}
@@ -142,29 +142,11 @@
int snd_oxfw_create_midi(struct snd_oxfw *oxfw)
{
- struct snd_oxfw_stream_formation formation;
struct snd_rawmidi *rmidi;
struct snd_rawmidi_str *str;
- u8 *format;
- int i, err;
+ int err;
- /* If its stream has MIDI conformant data channel, add one MIDI port */
- for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
- format = oxfw->tx_stream_formats[i];
- if (format != NULL) {
- err = snd_oxfw_stream_parse_format(format, &formation);
- if (err >= 0 && formation.midi > 0)
- oxfw->midi_input_ports = 1;
- }
-
- format = oxfw->rx_stream_formats[i];
- if (format != NULL) {
- err = snd_oxfw_stream_parse_format(format, &formation);
- if (err >= 0 && formation.midi > 0)
- oxfw->midi_output_ports = 1;
- }
- }
- if ((oxfw->midi_input_ports == 0) && (oxfw->midi_output_ports == 0))
+ if (oxfw->midi_input_ports == 0 && oxfw->midi_output_ports == 0)
return 0;
/* create midi ports */
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 9c73930..8d23341 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -134,11 +134,11 @@
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
stream = &oxfw->tx_stream;
formats = oxfw->tx_stream_formats;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
stream = &oxfw->rx_stream;
formats = oxfw->rx_stream_formats;
}
@@ -158,7 +158,7 @@
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
end:
return err;
}
@@ -244,7 +244,7 @@
mutex_unlock(&oxfw->mutex);
}
- amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
+ amdtp_am824_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
return 0;
}
@@ -265,7 +265,7 @@
mutex_unlock(&oxfw->mutex);
}
- amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
+ amdtp_am824_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 77ad5b9..7cb5743 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -148,14 +148,17 @@
}
pcm_channels = formation.pcm;
- midi_ports = DIV_ROUND_UP(formation.midi, 8);
+ midi_ports = formation.midi * 8;
/* The stream should have one pcm channels at least */
if (pcm_channels == 0) {
err = -EINVAL;
goto end;
}
- amdtp_stream_set_parameters(stream, rate, pcm_channels, midi_ports);
+ err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports,
+ false);
+ if (err < 0)
+ goto end;
err = cmp_connection_establish(conn,
amdtp_stream_get_max_payload(stream));
@@ -225,7 +228,7 @@
if (err < 0)
goto end;
- err = amdtp_stream_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
+ err = amdtp_am824_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
cmp_connection_destroy(conn);
@@ -238,9 +241,12 @@
* packets. As a result, next isochronous packet includes more data
* blocks than IEC 61883-6 defines.
*/
- if (stream == &oxfw->tx_stream)
+ if (stream == &oxfw->tx_stream) {
oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK |
CIP_JUMBO_PAYLOAD;
+ if (oxfw->wrong_dbs)
+ oxfw->tx_stream.flags |= CIP_WRONG_DBS;
+ }
end:
return err;
}
@@ -480,8 +486,8 @@
}
}
- if (formation->pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
- formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+ if (formation->pcm > AM824_MAX_CHANNELS_FOR_PCM ||
+ formation->midi > AM824_MAX_CHANNELS_FOR_MIDI)
return -ENOSYS;
return 0;
@@ -623,6 +629,9 @@
int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
{
u8 plugs[AVC_PLUG_INFO_BUF_BYTES];
+ struct snd_oxfw_stream_formation formation;
+ u8 *format;
+ unsigned int i;
int err;
/* the number of plugs for isoc in/out, ext in/out */
@@ -642,12 +651,42 @@
err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_OUT, 0);
if (err < 0)
goto end;
+
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+ format = oxfw->tx_stream_formats[i];
+ if (format == NULL)
+ continue;
+ err = snd_oxfw_stream_parse_format(format, &formation);
+ if (err < 0)
+ continue;
+
+ /* Add one MIDI port. */
+ if (formation.midi > 0)
+ oxfw->midi_input_ports = 1;
+ }
+
oxfw->has_output = true;
}
/* use iPCR[0] if exists */
- if (plugs[0] > 0)
+ if (plugs[0] > 0) {
err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0);
+ if (err < 0)
+ goto end;
+
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+ format = oxfw->rx_stream_formats[i];
+ if (format == NULL)
+ continue;
+ err = snd_oxfw_stream_parse_format(format, &formation);
+ if (err < 0)
+ continue;
+
+ /* Add one MIDI port. */
+ if (formation.midi > 0)
+ oxfw->midi_output_ports = 1;
+ }
+ }
end:
return err;
}
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 8c6ce01..588b93f 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -18,6 +18,9 @@
#define VENDOR_GRIFFIN 0x001292
#define VENDOR_BEHRINGER 0x001564
#define VENDOR_LACIE 0x00d04b
+#define VENDOR_TASCAM 0x00022e
+
+#define MODEL_SATELLITE 0x00200f
#define SPECIFIER_1394TA 0x00a02d
#define VERSION_AVC 0x010001
@@ -129,6 +132,40 @@
mutex_destroy(&oxfw->mutex);
}
+static void detect_quirks(struct snd_oxfw *oxfw)
+{
+ struct fw_device *fw_dev = fw_parent_device(oxfw->unit);
+ struct fw_csr_iterator it;
+ int key, val;
+ int vendor, model;
+
+ /* Seek from Root Directory of Config ROM. */
+ vendor = model = 0;
+ fw_csr_iterator_init(&it, fw_dev->config_rom + 5);
+ while (fw_csr_iterator_next(&it, &key, &val)) {
+ if (key == CSR_VENDOR)
+ vendor = val;
+ else if (key == CSR_MODEL)
+ model = val;
+ }
+
+ /*
+ * Mackie Onyx Satellite with base station has a quirk to report a wrong
+ * value in 'dbs' field of CIP header against its format information.
+ */
+ if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE)
+ oxfw->wrong_dbs = true;
+
+ /*
+ * TASCAM FireOne has physical control and requires a pair of additional
+ * MIDI ports.
+ */
+ if (vendor == VENDOR_TASCAM) {
+ oxfw->midi_input_ports++;
+ oxfw->midi_output_ports++;
+ }
+}
+
static int oxfw_probe(struct fw_unit *unit,
const struct ieee1394_device_id *id)
{
@@ -157,6 +194,8 @@
if (err < 0)
goto error;
+ detect_quirks(oxfw);
+
err = name_card(oxfw);
if (err < 0)
goto error;
@@ -294,6 +333,13 @@
.specifier_id = SPECIFIER_1394TA,
.version = VERSION_AVC,
},
+ /* TASCAM, FireOne */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = VENDOR_TASCAM,
+ .model_id = 0x800007,
+ },
{ }
};
MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table);
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index cace5ad..8392c42 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -28,7 +28,7 @@
#include "../fcp.h"
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
struct device_info {
@@ -49,6 +49,7 @@
struct mutex mutex;
spinlock_t lock;
+ bool wrong_dbs;
bool has_output;
u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile
new file mode 100644
index 0000000..0fc955d
--- /dev/null
+++ b/sound/firewire/tascam/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \
+ tascam-pcm.o tascam-hwdep.o tascam-transaction.o \
+ tascam-midi.o tascam.o
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
new file mode 100644
index 0000000..9dd0fcc
--- /dev/null
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -0,0 +1,243 @@
+/*
+ * amdtp-tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "tascam.h"
+
+#define AMDTP_FMT_TSCM_TX 0x1e
+#define AMDTP_FMT_TSCM_RX 0x3e
+
+struct amdtp_tscm {
+ unsigned int pcm_channels;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+};
+
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate)
+{
+ struct amdtp_tscm *p = s->protocol;
+ unsigned int data_channels;
+
+ if (amdtp_stream_running(s))
+ return -EBUSY;
+
+ data_channels = p->pcm_channels;
+
+ /* Packets in in-stream have extra 2 data channels. */
+ if (s->direction == AMDTP_IN_STREAM)
+ data_channels += 2;
+
+ return amdtp_stream_set_parameters(s, rate, data_channels);
+}
+
+static void write_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32(*src);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32(*src << 16);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ /* The first data channel is for event counter. */
+ buffer += 1;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[c]);
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_tscm *p = s->protocol;
+ unsigned int channels, i, c;
+
+ channels = p->pcm_channels;
+
+ for (i = 0; i < data_blocks; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[c] = 0x00000000;
+ buffer += s->data_block_quadlets;
+ }
+}
+
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ /*
+ * Our implementation allows this protocol to deliver 24 bit sample in
+ * 32bit data channel.
+ */
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+
+ return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_tscm *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+ struct snd_pcm_substream *pcm;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (data_blocks > 0 && pcm)
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+
+ /* A place holder for control messages. */
+
+ return data_blocks;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+ struct snd_pcm_substream *pcm;
+
+ /* This field is not used. */
+ *syt = 0x0000;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm)
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ else
+ write_pcm_silence(s, buffer, data_blocks);
+
+ return data_blocks;
+}
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, unsigned int pcm_channels)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+ struct amdtp_tscm *p;
+ unsigned int fmt;
+ int err;
+
+ if (dir == AMDTP_IN_STREAM) {
+ fmt = AMDTP_FMT_TSCM_TX;
+ process_data_blocks = process_tx_data_blocks;
+ } else {
+ fmt = AMDTP_FMT_TSCM_RX;
+ process_data_blocks = process_rx_data_blocks;
+ }
+
+ err = amdtp_stream_init(s, unit, dir,
+ CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt,
+ process_data_blocks, sizeof(struct amdtp_tscm));
+ if (err < 0)
+ return 0;
+
+ /* Use fixed value for FDF field. */
+ s->fdf = 0x00;
+
+ /* This protocol uses fixed number of data channels for PCM samples. */
+ p = s->protocol;
+ p->pcm_channels = pcm_channels;
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
new file mode 100644
index 0000000..131267c
--- /dev/null
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -0,0 +1,201 @@
+/*
+ * tascam-hwdep.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ */
+
+#include "tascam.h"
+
+static long hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
+ long count)
+{
+ union snd_firewire_event event;
+
+ memset(&event, 0, sizeof(event));
+
+ event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+ event.lock_status.status = (tscm->dev_lock_count > 0);
+ tscm->dev_lock_changed = false;
+
+ count = min_t(long, count, sizeof(event.lock_status));
+
+ if (copy_to_user(buf, &event, count))
+ return -EFAULT;
+
+ return count;
+}
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
+ loff_t *offset)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+ DEFINE_WAIT(wait);
+ union snd_firewire_event event;
+
+ spin_lock_irq(&tscm->lock);
+
+ while (!tscm->dev_lock_changed) {
+ prepare_to_wait(&tscm->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&tscm->lock);
+ schedule();
+ finish_wait(&tscm->hwdep_wait, &wait);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ spin_lock_irq(&tscm->lock);
+ }
+
+ memset(&event, 0, sizeof(event));
+ count = hwdep_read_locked(tscm, buf, count);
+ spin_unlock_irq(&tscm->lock);
+
+ return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+ poll_table *wait)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+ unsigned int events;
+
+ poll_wait(file, &tscm->hwdep_wait, wait);
+
+ spin_lock_irq(&tscm->lock);
+ if (tscm->dev_lock_changed)
+ events = POLLIN | POLLRDNORM;
+ else
+ events = 0;
+ spin_unlock_irq(&tscm->lock);
+
+ return events;
+}
+
+static int hwdep_get_info(struct snd_tscm *tscm, void __user *arg)
+{
+ struct fw_device *dev = fw_parent_device(tscm->unit);
+ struct snd_firewire_get_info info;
+
+ memset(&info, 0, sizeof(info));
+ info.type = SNDRV_FIREWIRE_TYPE_TASCAM;
+ info.card = dev->card->index;
+ *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+ *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+ strlcpy(info.device_name, dev_name(&dev->device),
+ sizeof(info.device_name));
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+static int hwdep_lock(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ if (tscm->dev_lock_count == 0) {
+ tscm->dev_lock_count = -1;
+ err = 0;
+ } else {
+ err = -EBUSY;
+ }
+
+ spin_unlock_irq(&tscm->lock);
+
+ return err;
+}
+
+static int hwdep_unlock(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ if (tscm->dev_lock_count == -1) {
+ tscm->dev_lock_count = 0;
+ err = 0;
+ } else {
+ err = -EBADFD;
+ }
+
+ spin_unlock_irq(&tscm->lock);
+
+ return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+
+ spin_lock_irq(&tscm->lock);
+ if (tscm->dev_lock_count == -1)
+ tscm->dev_lock_count = 0;
+ spin_unlock_irq(&tscm->lock);
+
+ return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+
+ switch (cmd) {
+ case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+ return hwdep_get_info(tscm, (void __user *)arg);
+ case SNDRV_FIREWIRE_IOCTL_LOCK:
+ return hwdep_lock(tscm);
+ case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+ return hwdep_unlock(tscm);
+ default:
+ return -ENOIOCTLCMD;
+ }
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hwdep_ioctl(hwdep, file, cmd,
+ (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+ .read = hwdep_read,
+ .release = hwdep_release,
+ .poll = hwdep_poll,
+ .ioctl = hwdep_ioctl,
+ .ioctl_compat = hwdep_compat_ioctl,
+};
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm)
+{
+ struct snd_hwdep *hwdep;
+ int err;
+
+ err = snd_hwdep_new(tscm->card, "Tascam", 0, &hwdep);
+ if (err < 0)
+ return err;
+
+ strcpy(hwdep->name, "Tascam");
+ hwdep->iface = SNDRV_HWDEP_IFACE_FW_TASCAM;
+ hwdep->ops = hwdep_ops;
+ hwdep->private_data = tscm;
+ hwdep->exclusive = true;
+
+ return err;
+}
diff --git a/sound/firewire/tascam/tascam-midi.c b/sound/firewire/tascam/tascam-midi.c
new file mode 100644
index 0000000..41f8420
--- /dev/null
+++ b/sound/firewire/tascam/tascam-midi.c
@@ -0,0 +1,135 @@
+/*
+ * tascam-midi.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static int midi_capture_open(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_playback_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+
+ /* Initialize internal status. */
+ tscm->running_status[substream->number] = 0;
+ tscm->on_sysex[substream->number] = 0;
+ return 0;
+}
+
+static int midi_capture_close(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_playback_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+
+ snd_fw_async_midi_port_finish(&tscm->out_ports[substream->number]);
+
+ return 0;
+}
+
+static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_tscm *tscm = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&tscm->lock, flags);
+
+ if (up)
+ tscm->tx_midi_substreams[substrm->number] = substrm;
+ else
+ tscm->tx_midi_substreams[substrm->number] = NULL;
+
+ spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_tscm *tscm = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&tscm->lock, flags);
+
+ if (up)
+ snd_fw_async_midi_port_run(&tscm->out_ports[substrm->number],
+ substrm);
+
+ spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_capture_ops = {
+ .open = midi_capture_open,
+ .close = midi_capture_close,
+ .trigger = midi_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_playback_ops = {
+ .open = midi_playback_open,
+ .close = midi_playback_close,
+ .trigger = midi_playback_trigger,
+};
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_str *stream;
+ struct snd_rawmidi_substream *subs;
+ int err;
+
+ err = snd_rawmidi_new(tscm->card, tscm->card->driver, 0,
+ tscm->spec->midi_playback_ports,
+ tscm->spec->midi_capture_ports,
+ &rmidi);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi->name, sizeof(rmidi->name),
+ "%s MIDI", tscm->card->shortname);
+ rmidi->private_data = tscm;
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_capture_ops);
+ stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+
+ /* Set port names for MIDI input. */
+ list_for_each_entry(subs, &stream->substreams, list) {
+ /* TODO: support virtual MIDI ports. */
+ if (subs->number < tscm->spec->midi_capture_ports) {
+ /* Hardware MIDI ports. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ tscm->card->shortname, subs->number + 1);
+ }
+ }
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_playback_ops);
+ stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+
+ /* Set port names for MIDI ourput. */
+ list_for_each_entry(subs, &stream->substreams, list) {
+ if (subs->number < tscm->spec->midi_playback_ports) {
+ /* Hardware MIDI ports only. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ tscm->card->shortname, subs->number + 1);
+ }
+ }
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
new file mode 100644
index 0000000..380d3db
--- /dev/null
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -0,0 +1,312 @@
+/*
+ * tascam-pcm.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static void set_buffer_params(struct snd_pcm_hardware *hw)
+{
+ hw->period_bytes_min = 4 * hw->channels_min;
+ hw->period_bytes_max = hw->period_bytes_min * 2048;
+ hw->buffer_bytes_max = hw->period_bytes_max * 2;
+
+ hw->periods_min = 2;
+ hw->periods_max = UINT_MAX;
+}
+
+static int pcm_init_hw_params(struct snd_tscm *tscm,
+ struct snd_pcm_substream *substream)
+{
+ static const struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 18,
+ };
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct amdtp_stream *stream;
+ unsigned int pcm_channels;
+
+ runtime->hw = hardware;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+ stream = &tscm->tx_stream;
+ pcm_channels = tscm->spec->pcm_capture_analog_channels;
+ } else {
+ runtime->hw.formats =
+ SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32;
+ stream = &tscm->rx_stream;
+ pcm_channels = tscm->spec->pcm_playback_analog_channels;
+ }
+
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels;
+
+ set_buffer_params(&runtime->hw);
+
+ return amdtp_tscm_add_pcm_hw_constraints(stream, runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ enum snd_tscm_clock clock;
+ unsigned int rate;
+ int err;
+
+ err = snd_tscm_stream_lock_try(tscm);
+ if (err < 0)
+ goto end;
+
+ err = pcm_init_hw_params(tscm, substream);
+ if (err < 0)
+ goto err_locked;
+
+ err = snd_tscm_stream_get_clock(tscm, &clock);
+ if (clock != SND_TSCM_CLOCK_INTERNAL ||
+ amdtp_stream_pcm_running(&tscm->rx_stream) ||
+ amdtp_stream_pcm_running(&tscm->tx_stream)) {
+ err = snd_tscm_stream_get_rate(tscm, &rate);
+ if (err < 0)
+ goto err_locked;
+ substream->runtime->hw.rate_min = rate;
+ substream->runtime->hw.rate_max = rate;
+ }
+
+ snd_pcm_set_sync(substream);
+end:
+ return err;
+err_locked:
+ snd_tscm_stream_lock_release(tscm);
+ return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ snd_tscm_stream_lock_release(tscm);
+
+ return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&tscm->mutex);
+ tscm->substreams_counter++;
+ mutex_unlock(&tscm->mutex);
+ }
+
+ amdtp_tscm_set_pcm_format(&tscm->tx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&tscm->mutex);
+ tscm->substreams_counter++;
+ mutex_unlock(&tscm->mutex);
+ }
+
+ amdtp_tscm_set_pcm_format(&tscm->rx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ mutex_lock(&tscm->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ tscm->substreams_counter--;
+
+ snd_tscm_stream_stop_duplex(tscm);
+
+ mutex_unlock(&tscm->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ mutex_lock(&tscm->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ tscm->substreams_counter--;
+
+ snd_tscm_stream_stop_duplex(tscm);
+
+ mutex_unlock(&tscm->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&tscm->mutex);
+
+ err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&tscm->tx_stream);
+
+ mutex_unlock(&tscm->mutex);
+
+ return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&tscm->mutex);
+
+ err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&tscm->rx_stream);
+
+ mutex_unlock(&tscm->mutex);
+
+ return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&tscm->tx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&tscm->tx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&tscm->rx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&tscm->rx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_tscm *tscm = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&tscm->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_tscm *tscm = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&tscm->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_capture_hw_params,
+ .hw_free = pcm_capture_hw_free,
+ .prepare = pcm_capture_prepare,
+ .trigger = pcm_capture_trigger,
+ .pointer = pcm_capture_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_playback_hw_params,
+ .hw_free = pcm_playback_hw_free,
+ .prepare = pcm_playback_prepare,
+ .trigger = pcm_playback_trigger,
+ .pointer = pcm_playback_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(tscm->card, tscm->card->driver, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = tscm;
+ snprintf(pcm->name, sizeof(pcm->name),
+ "%s PCM", tscm->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c
new file mode 100644
index 0000000..bfd4a4c
--- /dev/null
+++ b/sound/firewire/tascam/tascam-proc.c
@@ -0,0 +1,88 @@
+/*
+ * tascam-proc.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "./tascam.h"
+
+static void proc_read_firmware(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_tscm *tscm = entry->private_data;
+ __be32 data;
+ unsigned int reg, fpga, arm, hw;
+ int err;
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_REGISTER,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ reg = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_FPGA,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ fpga = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_ARM,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ arm = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_HW,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ hw = be32_to_cpu(data);
+
+ snd_iprintf(buffer, "Register: %d (0x%08x)\n", reg & 0xffff, reg);
+ snd_iprintf(buffer, "FPGA: %d (0x%08x)\n", fpga & 0xffff, fpga);
+ snd_iprintf(buffer, "ARM: %d (0x%08x)\n", arm & 0xffff, arm);
+ snd_iprintf(buffer, "Hardware: %d (0x%08x)\n", hw >> 16, hw);
+}
+
+static void add_node(struct snd_tscm *tscm, struct snd_info_entry *root,
+ const char *name,
+ void (*op)(struct snd_info_entry *e,
+ struct snd_info_buffer *b))
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_card_entry(tscm->card, name, root);
+ if (entry == NULL)
+ return;
+
+ snd_info_set_text_ops(entry, tscm, op);
+ if (snd_info_register(entry) < 0)
+ snd_info_free_entry(entry);
+}
+
+void snd_tscm_proc_init(struct snd_tscm *tscm)
+{
+ struct snd_info_entry *root;
+
+ /*
+ * All nodes are automatically removed at snd_card_disconnect(),
+ * by following to link list.
+ */
+ root = snd_info_create_card_entry(tscm->card, "firewire",
+ tscm->card->proc_root);
+ if (root == NULL)
+ return;
+ root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(root) < 0) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ add_node(tscm, root, "firmware", proc_read_firmware);
+}
diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
new file mode 100644
index 0000000..0e6dd5c6
--- /dev/null
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -0,0 +1,496 @@
+/*
+ * tascam-stream.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/delay.h>
+#include "tascam.h"
+
+#define CALLBACK_TIMEOUT 500
+
+static int get_clock(struct snd_tscm *tscm, u32 *data)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+ ®, sizeof(reg), 0);
+ if (err >= 0)
+ *data = be32_to_cpu(reg);
+
+ return err;
+}
+
+static int set_clock(struct snd_tscm *tscm, unsigned int rate,
+ enum snd_tscm_clock clock)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+ data &= 0x0000ffff;
+
+ if (rate > 0) {
+ data &= 0x000000ff;
+ /* Base rate. */
+ if ((rate % 44100) == 0) {
+ data |= 0x00000100;
+ /* Multiplier. */
+ if (rate / 44100 == 2)
+ data |= 0x00008000;
+ } else if ((rate % 48000) == 0) {
+ data |= 0x00000200;
+ /* Multiplier. */
+ if (rate / 48000 == 2)
+ data |= 0x00008000;
+ } else {
+ return -EAGAIN;
+ }
+ }
+
+ if (clock != INT_MAX) {
+ data &= 0x0000ff00;
+ data |= clock + 1;
+ }
+
+ reg = cpu_to_be32(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ if (data & 0x00008000)
+ reg = cpu_to_be32(0x0000001a);
+ else
+ reg = cpu_to_be32(0x0000000d);
+
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MULTIPLEX_MODE,
+ ®, sizeof(reg), 0);
+}
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate)
+{
+ u32 data = 0x0;
+ unsigned int trials = 0;
+ int err;
+
+ while (data == 0x0 || trials++ < 5) {
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+
+ data = (data & 0xff000000) >> 24;
+ }
+
+ /* Check base rate. */
+ if ((data & 0x0f) == 0x01)
+ *rate = 44100;
+ else if ((data & 0x0f) == 0x02)
+ *rate = 48000;
+ else
+ return -EAGAIN;
+
+ /* Check multiplier. */
+ if ((data & 0xf0) == 0x80)
+ *rate *= 2;
+ else if ((data & 0xf0) != 0x00)
+ return -EAGAIN;
+
+ return err;
+}
+
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm, enum snd_tscm_clock *clock)
+{
+ u32 data;
+ int err;
+
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+
+ *clock = ((data & 0x00ff0000) >> 16) - 1;
+ if (*clock < 0 || *clock > SND_TSCM_CLOCK_ADAT)
+ return -EIO;
+
+ return 0;
+}
+
+static int enable_data_channels(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ u32 data;
+ unsigned int i;
+ int err;
+
+ data = 0;
+ for (i = 0; i < tscm->spec->pcm_capture_analog_channels; ++i)
+ data |= BIT(i);
+ if (tscm->spec->has_adat)
+ data |= 0x0000ff00;
+ if (tscm->spec->has_spdif)
+ data |= 0x00030000;
+
+ reg = cpu_to_be32(data);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_TX_PCM_CHANNELS,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = 0;
+ for (i = 0; i < tscm->spec->pcm_playback_analog_channels; ++i)
+ data |= BIT(i);
+ if (tscm->spec->has_adat)
+ data |= 0x0000ff00;
+ if (tscm->spec->has_spdif)
+ data |= 0x00030000;
+
+ reg = cpu_to_be32(data);
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_RX_PCM_CHANNELS,
+ ®, sizeof(reg), 0);
+}
+
+static int set_stream_formats(struct snd_tscm *tscm, unsigned int rate)
+{
+ __be32 reg;
+ int err;
+
+ /* Set an option for unknown purpose. */
+ reg = cpu_to_be32(0x00200000);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ err = enable_data_channels(tscm);
+ if (err < 0)
+ return err;
+
+ return set_clock(tscm, rate, INT_MAX);
+}
+
+static void finish_session(struct snd_tscm *tscm)
+{
+ __be32 reg;
+
+ reg = 0;
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+ ®, sizeof(reg), 0);
+
+ reg = 0;
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+ ®, sizeof(reg), 0);
+
+}
+
+static int begin_session(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ int err;
+
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Set an option for unknown purpose. */
+ reg = cpu_to_be32(0x00002000);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Start multiplexing PCM samples on packets. */
+ reg = cpu_to_be32(0x00000001);
+ return snd_fw_transaction(tscm->unit,
+ TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_ON,
+ ®, sizeof(reg), 0);
+}
+
+static void release_resources(struct snd_tscm *tscm)
+{
+ __be32 reg;
+
+ /* Unregister channels. */
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+ ®, sizeof(reg), 0);
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+ ®, sizeof(reg), 0);
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+ ®, sizeof(reg), 0);
+
+ /* Release isochronous resources. */
+ fw_iso_resources_free(&tscm->tx_resources);
+ fw_iso_resources_free(&tscm->rx_resources);
+}
+
+static int keep_resources(struct snd_tscm *tscm, unsigned int rate)
+{
+ __be32 reg;
+ int err;
+
+ /* Keep resources for in-stream. */
+ err = amdtp_tscm_set_parameters(&tscm->tx_stream, rate);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&tscm->tx_resources,
+ amdtp_stream_get_max_payload(&tscm->tx_stream),
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ /* Keep resources for out-stream. */
+ err = amdtp_tscm_set_parameters(&tscm->rx_stream, rate);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&tscm->rx_resources,
+ amdtp_stream_get_max_payload(&tscm->rx_stream),
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ return err;
+
+ /* Register the isochronous channel for transmitting stream. */
+ reg = cpu_to_be32(tscm->tx_resources.channel);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ /* Unknown */
+ reg = cpu_to_be32(0x00000002);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ /* Register the isochronous channel for receiving stream. */
+ reg = cpu_to_be32(tscm->rx_resources.channel);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ release_resources(tscm);
+ return err;
+}
+
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
+{
+ unsigned int pcm_channels;
+ int err;
+
+ /* For out-stream. */
+ err = fw_iso_resources_init(&tscm->rx_resources, tscm->unit);
+ if (err < 0)
+ return err;
+ pcm_channels = tscm->spec->pcm_playback_analog_channels;
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ err = amdtp_tscm_init(&tscm->rx_stream, tscm->unit, AMDTP_OUT_STREAM,
+ pcm_channels);
+ if (err < 0)
+ return err;
+
+ /* For in-stream. */
+ err = fw_iso_resources_init(&tscm->tx_resources, tscm->unit);
+ if (err < 0)
+ return err;
+ pcm_channels = tscm->spec->pcm_capture_analog_channels;
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ err = amdtp_tscm_init(&tscm->tx_stream, tscm->unit, AMDTP_IN_STREAM,
+ pcm_channels);
+ if (err < 0)
+ amdtp_stream_destroy(&tscm->rx_stream);
+
+ return 0;
+}
+
+/* At bus reset, streaming is stopped and some registers are clear. */
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm)
+{
+ amdtp_stream_pcm_abort(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->tx_stream);
+
+ amdtp_stream_pcm_abort(&tscm->rx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm)
+{
+ amdtp_stream_destroy(&tscm->rx_stream);
+ amdtp_stream_destroy(&tscm->tx_stream);
+
+ fw_iso_resources_destroy(&tscm->rx_resources);
+ fw_iso_resources_destroy(&tscm->tx_resources);
+}
+
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
+{
+ unsigned int curr_rate;
+ int err;
+
+ if (tscm->substreams_counter == 0)
+ return 0;
+
+ err = snd_tscm_stream_get_rate(tscm, &curr_rate);
+ if (err < 0)
+ return err;
+ if (curr_rate != rate ||
+ amdtp_streaming_error(&tscm->tx_stream) ||
+ amdtp_streaming_error(&tscm->rx_stream)) {
+ finish_session(tscm);
+
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ release_resources(tscm);
+ }
+
+ if (!amdtp_stream_running(&tscm->tx_stream)) {
+ amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE,
+ &tscm->tx_stream, &tscm->rx_stream);
+ err = keep_resources(tscm, rate);
+ if (err < 0)
+ goto error;
+
+ err = set_stream_formats(tscm, rate);
+ if (err < 0)
+ goto error;
+
+ err = begin_session(tscm);
+ if (err < 0)
+ goto error;
+
+ err = amdtp_stream_start(&tscm->tx_stream,
+ tscm->tx_resources.channel,
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&tscm->tx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ if (!amdtp_stream_running(&tscm->rx_stream)) {
+ err = amdtp_stream_start(&tscm->rx_stream,
+ tscm->rx_resources.channel,
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&tscm->rx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ return 0;
+error:
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ finish_session(tscm);
+ release_resources(tscm);
+
+ return err;
+}
+
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm)
+{
+ if (tscm->substreams_counter > 0)
+ return;
+
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ finish_session(tscm);
+ release_resources(tscm);
+}
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm)
+{
+ tscm->dev_lock_changed = true;
+ wake_up(&tscm->hwdep_wait);
+}
+
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ /* user land lock this */
+ if (tscm->dev_lock_count < 0) {
+ err = -EBUSY;
+ goto end;
+ }
+
+ /* this is the first time */
+ if (tscm->dev_lock_count++ == 0)
+ snd_tscm_stream_lock_changed(tscm);
+ err = 0;
+end:
+ spin_unlock_irq(&tscm->lock);
+ return err;
+}
+
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm)
+{
+ spin_lock_irq(&tscm->lock);
+
+ if (WARN_ON(tscm->dev_lock_count <= 0))
+ goto end;
+ if (--tscm->dev_lock_count == 0)
+ snd_tscm_stream_lock_changed(tscm);
+end:
+ spin_unlock_irq(&tscm->lock);
+}
diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c
new file mode 100644
index 0000000..904ce03
--- /dev/null
+++ b/sound/firewire/tascam/tascam-transaction.c
@@ -0,0 +1,302 @@
+/*
+ * tascam-transaction.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+/*
+ * When return minus value, given argument is not MIDI status.
+ * When return 0, given argument is a beginning of system exclusive.
+ * When return the others, given argument is MIDI data.
+ */
+static inline int calculate_message_bytes(u8 status)
+{
+ switch (status) {
+ case 0xf6: /* Tune request. */
+ case 0xf8: /* Timing clock. */
+ case 0xfa: /* Start. */
+ case 0xfb: /* Continue. */
+ case 0xfc: /* Stop. */
+ case 0xfe: /* Active sensing. */
+ case 0xff: /* System reset. */
+ return 1;
+ case 0xf1: /* MIDI time code quarter frame. */
+ case 0xf3: /* Song select. */
+ return 2;
+ case 0xf2: /* Song position pointer. */
+ return 3;
+ case 0xf0: /* Exclusive. */
+ return 0;
+ case 0xf7: /* End of exclusive. */
+ break;
+ case 0xf4: /* Undefined. */
+ case 0xf5: /* Undefined. */
+ case 0xf9: /* Undefined. */
+ case 0xfd: /* Undefined. */
+ break;
+ default:
+ switch (status & 0xf0) {
+ case 0x80: /* Note on. */
+ case 0x90: /* Note off. */
+ case 0xa0: /* Polyphonic key pressure. */
+ case 0xb0: /* Control change and Mode change. */
+ case 0xe0: /* Pitch bend change. */
+ return 3;
+ case 0xc0: /* Program change. */
+ case 0xd0: /* Channel pressure. */
+ return 2;
+ default:
+ break;
+ }
+ break;
+ }
+
+ return -EINVAL;
+}
+
+static int fill_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+ unsigned int port = substream->number;
+ int i, len, consume;
+ u8 *label, *msg;
+ u8 status;
+
+ /* The first byte is used for label, the rest for MIDI bytes. */
+ label = buf;
+ msg = buf + 1;
+
+ consume = snd_rawmidi_transmit_peek(substream, msg, 3);
+ if (consume == 0)
+ return 0;
+
+ /* On exclusive message. */
+ if (tscm->on_sysex[port]) {
+ /* Seek the end of exclusives. */
+ for (i = 0; i < consume; ++i) {
+ if (msg[i] == 0xf7) {
+ tscm->on_sysex[port] = false;
+ break;
+ }
+ }
+
+ /* At the end of exclusive message, use label 0x07. */
+ if (!tscm->on_sysex[port]) {
+ consume = i + 1;
+ *label = (port << 4) | 0x07;
+ /* During exclusive message, use label 0x04. */
+ } else if (consume == 3) {
+ *label = (port << 4) | 0x04;
+ /* We need to fill whole 3 bytes. Go to next change. */
+ } else {
+ return 0;
+ }
+
+ len = consume;
+ } else {
+ /* The beginning of exclusives. */
+ if (msg[0] == 0xf0) {
+ /* Transfer it in next chance in another condition. */
+ tscm->on_sysex[port] = true;
+ return 0;
+ } else {
+ /* On running-status. */
+ if ((msg[0] & 0x80) != 0x80)
+ status = tscm->running_status[port];
+ else
+ status = msg[0];
+
+ /* Calculate consume bytes. */
+ len = calculate_message_bytes(status);
+ if (len <= 0)
+ return 0;
+
+ /* On running-status. */
+ if ((msg[0] & 0x80) != 0x80) {
+ /* Enough MIDI bytes were not retrieved. */
+ if (consume < len - 1)
+ return 0;
+ consume = len - 1;
+
+ msg[2] = msg[1];
+ msg[1] = msg[0];
+ msg[0] = tscm->running_status[port];
+ } else {
+ /* Enough MIDI bytes were not retrieved. */
+ if (consume < len)
+ return 0;
+ consume = len;
+
+ tscm->running_status[port] = msg[0];
+ }
+ }
+
+ *label = (port << 4) | (msg[0] >> 4);
+ }
+
+ if (len > 0 && len < 3)
+ memset(msg + len, 0, 3 - len);
+
+ return consume;
+}
+
+static void handle_midi_tx(struct fw_card *card, struct fw_request *request,
+ int tcode, int destination, int source,
+ int generation, unsigned long long offset,
+ void *data, size_t length, void *callback_data)
+{
+ struct snd_tscm *tscm = callback_data;
+ u32 *buf = (u32 *)data;
+ unsigned int messages;
+ unsigned int i;
+ unsigned int port;
+ struct snd_rawmidi_substream *substream;
+ u8 *b;
+ int bytes;
+
+ if (offset != tscm->async_handler.offset)
+ goto end;
+
+ messages = length / 8;
+ for (i = 0; i < messages; i++) {
+ b = (u8 *)(buf + i * 2);
+
+ port = b[0] >> 4;
+ /* TODO: support virtual MIDI ports. */
+ if (port >= tscm->spec->midi_capture_ports)
+ goto end;
+
+ /* Assume the message length. */
+ bytes = calculate_message_bytes(b[1]);
+ /* On MIDI data or exclusives. */
+ if (bytes <= 0) {
+ /* Seek the end of exclusives. */
+ for (bytes = 1; bytes < 4; bytes++) {
+ if (b[bytes] == 0xf7)
+ break;
+ }
+ if (bytes == 4)
+ bytes = 3;
+ }
+
+ substream = ACCESS_ONCE(tscm->tx_midi_substreams[port]);
+ if (substream != NULL)
+ snd_rawmidi_receive(substream, b + 1, bytes);
+ }
+end:
+ fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm)
+{
+ static const struct fw_address_region resp_register_region = {
+ .start = 0xffffe0000000ull,
+ .end = 0xffffe000ffffull,
+ };
+ unsigned int i;
+ int err;
+
+ /*
+ * Usually, two quadlets are transferred by one transaction. The first
+ * quadlet has MIDI messages, the rest includes timestamp.
+ * Sometimes, 8 set of the data is transferred by a block transaction.
+ */
+ tscm->async_handler.length = 8 * 8;
+ tscm->async_handler.address_callback = handle_midi_tx;
+ tscm->async_handler.callback_data = tscm;
+
+ err = fw_core_add_address_handler(&tscm->async_handler,
+ &resp_register_region);
+ if (err < 0)
+ return err;
+
+ err = snd_tscm_transaction_reregister(tscm);
+ if (err < 0)
+ goto error;
+
+ for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++) {
+ err = snd_fw_async_midi_port_init(
+ &tscm->out_ports[i], tscm->unit,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_RX_QUAD,
+ 4, fill_message);
+ if (err < 0)
+ goto error;
+ }
+
+ return err;
+error:
+ fw_core_remove_address_handler(&tscm->async_handler);
+ return err;
+}
+
+/* At bus reset, these registers are cleared. */
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm)
+{
+ struct fw_device *device = fw_parent_device(tscm->unit);
+ __be32 reg;
+ int err;
+
+ /* Register messaging address. Block transaction is not allowed. */
+ reg = cpu_to_be32((device->card->node_id << 16) |
+ (tscm->async_handler.offset >> 32));
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ reg = cpu_to_be32(tscm->async_handler.offset);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Turn on messaging. */
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+ ®, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Turn on FireWire LED. */
+ reg = cpu_to_be32(0x0001008e);
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+ ®, sizeof(reg), 0);
+}
+
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ unsigned int i;
+
+ /* Turn off FireWire LED. */
+ reg = cpu_to_be32(0x0000008e);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+ ®, sizeof(reg), 0);
+
+ /* Turn off messaging. */
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+ ®, sizeof(reg), 0);
+
+ /* Unregister the address. */
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+ ®, sizeof(reg), 0);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+ ®, sizeof(reg), 0);
+
+ fw_core_remove_address_handler(&tscm->async_handler);
+ for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++)
+ snd_fw_async_midi_port_destroy(&tscm->out_ports[i]);
+}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
new file mode 100644
index 0000000..ee0bc18
--- /dev/null
+++ b/sound/firewire/tascam/tascam.c
@@ -0,0 +1,209 @@
+/*
+ * tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+MODULE_DESCRIPTION("TASCAM FireWire series Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+static struct snd_tscm_spec model_specs[] = {
+ {
+ .name = "FW-1884",
+ .has_adat = true,
+ .has_spdif = true,
+ .pcm_capture_analog_channels = 8,
+ .pcm_playback_analog_channels = 8,
+ .midi_capture_ports = 4,
+ .midi_playback_ports = 4,
+ .is_controller = true,
+ },
+ {
+ .name = "FW-1082",
+ .has_adat = false,
+ .has_spdif = true,
+ .pcm_capture_analog_channels = 8,
+ .pcm_playback_analog_channels = 2,
+ .midi_capture_ports = 2,
+ .midi_playback_ports = 2,
+ .is_controller = true,
+ },
+ /* FW-1804 may be supported. */
+};
+
+static int identify_model(struct snd_tscm *tscm)
+{
+ struct fw_device *fw_dev = fw_parent_device(tscm->unit);
+ const u32 *config_rom = fw_dev->config_rom;
+ char model[9];
+ unsigned int i;
+ u8 c;
+
+ if (fw_dev->config_rom_length < 30) {
+ dev_err(&tscm->unit->device,
+ "Configuration ROM is too short.\n");
+ return -ENODEV;
+ }
+
+ /* Pick up model name from certain addresses. */
+ for (i = 0; i < 8; i++) {
+ c = config_rom[28 + i / 4] >> (24 - 8 * (i % 4));
+ if (c == '\0')
+ break;
+ model[i] = c;
+ }
+ model[i] = '\0';
+
+ for (i = 0; i < ARRAY_SIZE(model_specs); i++) {
+ if (strcmp(model, model_specs[i].name) == 0) {
+ tscm->spec = &model_specs[i];
+ break;
+ }
+ }
+ if (tscm->spec == NULL)
+ return -ENODEV;
+
+ strcpy(tscm->card->driver, "FW-TASCAM");
+ strcpy(tscm->card->shortname, model);
+ strcpy(tscm->card->mixername, model);
+ snprintf(tscm->card->longname, sizeof(tscm->card->longname),
+ "TASCAM %s, GUID %08x%08x at %s, S%d", model,
+ fw_dev->config_rom[3], fw_dev->config_rom[4],
+ dev_name(&tscm->unit->device), 100 << fw_dev->max_speed);
+
+ return 0;
+}
+
+static void tscm_card_free(struct snd_card *card)
+{
+ struct snd_tscm *tscm = card->private_data;
+
+ snd_tscm_transaction_unregister(tscm);
+ snd_tscm_stream_destroy_duplex(tscm);
+
+ fw_unit_put(tscm->unit);
+
+ mutex_destroy(&tscm->mutex);
+}
+
+static int snd_tscm_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_card *card;
+ struct snd_tscm *tscm;
+ int err;
+
+ /* create card */
+ err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+ sizeof(struct snd_tscm), &card);
+ if (err < 0)
+ return err;
+ card->private_free = tscm_card_free;
+
+ /* initialize myself */
+ tscm = card->private_data;
+ tscm->card = card;
+ tscm->unit = fw_unit_get(unit);
+
+ mutex_init(&tscm->mutex);
+ spin_lock_init(&tscm->lock);
+ init_waitqueue_head(&tscm->hwdep_wait);
+
+ err = identify_model(tscm);
+ if (err < 0)
+ goto error;
+
+ snd_tscm_proc_init(tscm);
+
+ err = snd_tscm_stream_init_duplex(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_pcm_devices(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_transaction_register(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_midi_devices(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_hwdep_device(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ dev_set_drvdata(&unit->device, tscm);
+
+ return err;
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static void snd_tscm_update(struct fw_unit *unit)
+{
+ struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+ snd_tscm_transaction_reregister(tscm);
+
+ mutex_lock(&tscm->mutex);
+ snd_tscm_stream_update_duplex(tscm);
+ mutex_unlock(&tscm->mutex);
+}
+
+static void snd_tscm_remove(struct fw_unit *unit)
+{
+ struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(tscm->card);
+}
+
+static const struct ieee1394_device_id snd_tscm_id_table[] = {
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ },
+ /* FE-08 requires reverse-engineering because it just has faders. */
+ {}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
+
+static struct fw_driver tscm_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "snd-firewire-tascam",
+ .bus = &fw_bus_type,
+ },
+ .probe = snd_tscm_probe,
+ .update = snd_tscm_update,
+ .remove = snd_tscm_remove,
+ .id_table = snd_tscm_id_table,
+};
+
+static int __init snd_tscm_init(void)
+{
+ return driver_register(&tscm_driver.driver);
+}
+
+static void __exit snd_tscm_exit(void)
+{
+ driver_unregister(&tscm_driver.driver);
+}
+
+module_init(snd_tscm_init);
+module_exit(snd_tscm_exit);
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
new file mode 100644
index 0000000..2d028d2
--- /dev/null
+++ b/sound/firewire/tascam/tascam.h
@@ -0,0 +1,147 @@
+/*
+ * tascam.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_TASCAM_H_INCLUDED
+#define SOUND_TASCAM_H_INCLUDED
+
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/mutex.h>
+#include <linux/slab.h>
+#include <linux/compat.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../amdtp-stream.h"
+#include "../iso-resources.h"
+
+struct snd_tscm_spec {
+ const char *const name;
+ bool has_adat;
+ bool has_spdif;
+ unsigned int pcm_capture_analog_channels;
+ unsigned int pcm_playback_analog_channels;
+ unsigned int midi_capture_ports;
+ unsigned int midi_playback_ports;
+ bool is_controller;
+};
+
+#define TSCM_MIDI_IN_PORT_MAX 4
+#define TSCM_MIDI_OUT_PORT_MAX 4
+
+struct snd_tscm {
+ struct snd_card *card;
+ struct fw_unit *unit;
+
+ struct mutex mutex;
+ spinlock_t lock;
+
+ const struct snd_tscm_spec *spec;
+
+ struct fw_iso_resources tx_resources;
+ struct fw_iso_resources rx_resources;
+ struct amdtp_stream tx_stream;
+ struct amdtp_stream rx_stream;
+ unsigned int substreams_counter;
+
+ int dev_lock_count;
+ bool dev_lock_changed;
+ wait_queue_head_t hwdep_wait;
+
+ /* For MIDI message incoming transactions. */
+ struct fw_address_handler async_handler;
+ struct snd_rawmidi_substream *tx_midi_substreams[TSCM_MIDI_IN_PORT_MAX];
+
+ /* For MIDI message outgoing transactions. */
+ struct snd_fw_async_midi_port out_ports[TSCM_MIDI_OUT_PORT_MAX];
+ u8 running_status[TSCM_MIDI_OUT_PORT_MAX];
+ bool on_sysex[TSCM_MIDI_OUT_PORT_MAX];
+
+ /* For control messages. */
+ struct snd_firewire_tascam_status *status;
+};
+
+#define TSCM_ADDR_BASE 0xffff00000000ull
+
+#define TSCM_OFFSET_FIRMWARE_REGISTER 0x0000
+#define TSCM_OFFSET_FIRMWARE_FPGA 0x0004
+#define TSCM_OFFSET_FIRMWARE_ARM 0x0008
+#define TSCM_OFFSET_FIRMWARE_HW 0x000c
+
+#define TSCM_OFFSET_ISOC_TX_CH 0x0200
+#define TSCM_OFFSET_UNKNOWN 0x0204
+#define TSCM_OFFSET_START_STREAMING 0x0208
+#define TSCM_OFFSET_ISOC_RX_CH 0x020c
+#define TSCM_OFFSET_ISOC_RX_ON 0x0210 /* Little conviction. */
+#define TSCM_OFFSET_TX_PCM_CHANNELS 0x0214
+#define TSCM_OFFSET_RX_PCM_CHANNELS 0x0218
+#define TSCM_OFFSET_MULTIPLEX_MODE 0x021c
+#define TSCM_OFFSET_ISOC_TX_ON 0x0220
+/* Unknown 0x0224 */
+#define TSCM_OFFSET_CLOCK_STATUS 0x0228
+#define TSCM_OFFSET_SET_OPTION 0x022c
+
+#define TSCM_OFFSET_MIDI_TX_ON 0x0300
+#define TSCM_OFFSET_MIDI_TX_ADDR_HI 0x0304
+#define TSCM_OFFSET_MIDI_TX_ADDR_LO 0x0308
+
+#define TSCM_OFFSET_LED_POWER 0x0404
+
+#define TSCM_OFFSET_MIDI_RX_QUAD 0x4000
+
+enum snd_tscm_clock {
+ SND_TSCM_CLOCK_INTERNAL = 0,
+ SND_TSCM_CLOCK_WORD = 1,
+ SND_TSCM_CLOCK_SPDIF = 2,
+ SND_TSCM_CLOCK_ADAT = 3,
+};
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, unsigned int pcm_channels);
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate);
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate);
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm,
+ enum snd_tscm_clock *clock);
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm);
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate);
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm);
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm);
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm);
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm);
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm);
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm);
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm);
+
+void snd_tscm_proc_init(struct snd_tscm *tscm);
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm);
+
+#endif
diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c
index 33ba77d..cb89ec7 100644
--- a/sound/hda/ext/hdac_ext_stream.c
+++ b/sound/hda/ext/hdac_ext_stream.c
@@ -227,7 +227,7 @@
void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
int stream)
{
- snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 0);
+ snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 1 << stream);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_link_set_stream_id);
@@ -385,14 +385,13 @@
break;
case HDAC_EXT_STREAM_TYPE_HOST:
- if (stream->decoupled) {
+ if (stream->decoupled && !stream->link_locked)
snd_hdac_ext_stream_decouple(ebus, stream, false);
- snd_hdac_stream_release(&stream->hstream);
- }
+ snd_hdac_stream_release(&stream->hstream);
break;
case HDAC_EXT_STREAM_TYPE_LINK:
- if (stream->decoupled)
+ if (stream->decoupled && !stream->hstream.opened)
snd_hdac_ext_stream_decouple(ebus, stream, false);
spin_lock_irq(&bus->reg_lock);
stream->link_locked = 0;
diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c
index 89c2711..3060e2a 100644
--- a/sound/hda/hda_bus_type.c
+++ b/sound/hda/hda_bus_type.c
@@ -4,6 +4,7 @@
#include <linux/init.h>
#include <linux/device.h>
#include <linux/module.h>
+#include <linux/mod_devicetable.h>
#include <linux/export.h>
#include <sound/hdaudio.h>
@@ -63,9 +64,21 @@
return 1;
}
+static int hda_uevent(struct device *dev, struct kobj_uevent_env *env)
+{
+ char modalias[32];
+
+ snd_hdac_codec_modalias(dev_to_hdac_dev(dev), modalias,
+ sizeof(modalias));
+ if (add_uevent_var(env, "MODALIAS=%s", modalias))
+ return -ENOMEM;
+ return 0;
+}
+
struct bus_type snd_hda_bus_type = {
.name = "hdaudio",
.match = hda_bus_match,
+ .uevent = hda_uevent,
};
EXPORT_SYMBOL_GPL(snd_hda_bus_type);
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 27c447e..0e81ea89 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -172,6 +172,15 @@
}
}
+/**
+ * snd_hdac_bus_add_device - Add a codec to bus
+ * @bus: HDA core bus
+ * @codec: HDA core device to add
+ *
+ * Adds the given codec to the list in the bus. The caddr_tbl array
+ * and codec_powered bits are updated, as well.
+ * Returns zero if success, or a negative error code.
+ */
int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec)
{
if (bus->caddr_tbl[codec->addr]) {
@@ -188,6 +197,11 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device);
+/**
+ * snd_hdac_bus_remove_device - Remove a codec from bus
+ * @bus: HDA core bus
+ * @codec: HDA core device to remove
+ */
void snd_hdac_bus_remove_device(struct hdac_bus *bus,
struct hdac_device *codec)
{
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index db96042..e361024 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -164,6 +164,43 @@
EXPORT_SYMBOL_GPL(snd_hdac_device_unregister);
/**
+ * snd_hdac_device_set_chip_name - set/update the codec name
+ * @codec: the HDAC device
+ * @name: name string to set
+ *
+ * Returns 0 if the name is set or updated, or a negative error code.
+ */
+int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name)
+{
+ char *newname;
+
+ if (!name)
+ return 0;
+ newname = kstrdup(name, GFP_KERNEL);
+ if (!newname)
+ return -ENOMEM;
+ kfree(codec->chip_name);
+ codec->chip_name = newname;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name);
+
+/**
+ * snd_hdac_codec_modalias - give the module alias name
+ * @codec: HDAC device
+ * @buf: string buffer to store
+ * @size: string buffer size
+ *
+ * Returns the size of string, like snprintf(), or a negative error code.
+ */
+int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size)
+{
+ return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
+ codec->vendor_id, codec->revision_id, codec->type);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);
+
+/**
* snd_hdac_make_cmd - compose a 32bit command word to be sent to the
* HD-audio controller
* @codec: the codec object
@@ -592,8 +629,10 @@
EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm);
#endif
-/*
- * Enable/disable the link power for a codec.
+/**
+ * snd_hdac_link_power - Enable/disable the link power for a codec
+ * @codec: the codec object
+ * @bool: enable or disable the link power
*/
int snd_hdac_link_power(struct hdac_device *codec, bool enable)
{
@@ -952,3 +991,84 @@
return true;
}
EXPORT_SYMBOL_GPL(snd_hdac_is_supported_format);
+
+static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+ unsigned int res;
+
+ if (snd_hdac_exec_verb(hdac, cmd, flags, &res))
+ return -1;
+
+ return res;
+}
+
+static int codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+
+ return snd_hdac_exec_verb(hdac, cmd, flags, NULL);
+}
+
+/**
+ * snd_hdac_codec_read - send a command and get the response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command and read the corresponding response.
+ *
+ * Returns the obtained response value, or -1 for an error.
+ */
+int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ return codec_read(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_read);
+
+/**
+ * snd_hdac_codec_write - send a single command without waiting for response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ return codec_write(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_write);
+
+/**
+ * snd_hdac_check_power_state - check whether the actual power state matches
+ * with the target state
+ *
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @target_state: target state to check for
+ *
+ * Return true if state matches, false if not
+ */
+bool snd_hdac_check_power_state(struct hdac_device *hdac,
+ hda_nid_t nid, unsigned int target_state)
+{
+ unsigned int state = codec_read(hdac, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+
+ if (state & AC_PWRST_ERROR)
+ return true;
+ state = (state >> 4) & 0x0f;
+ return (state == target_state);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_check_power_state);
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 55c3df4..8fef1b8 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -23,6 +23,19 @@
static struct i915_audio_component *hdac_acomp;
+/**
+ * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup
+ * @bus: HDA core bus
+ * @enable: enable or disable the wakeup
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function should be called during the chip reset, also called at
+ * resume for updating STATESTS register read.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
{
struct i915_audio_component *acomp = bus->audio_component;
@@ -45,6 +58,19 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup);
+/**
+ * snd_hdac_display_power - Power up / down the power refcount
+ * @bus: HDA core bus
+ * @enable: power up or down
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function manages a refcount and calls the i915 get_power() and
+ * put_power() ops accordingly, toggling the codec wakeup, too.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
{
struct i915_audio_component *acomp = bus->audio_component;
@@ -71,6 +97,16 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_display_power);
+/**
+ * snd_hdac_get_display_clk - Get CDCLK in kHz
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function queries CDCLK value in kHz from the graphics driver and
+ * returns the value. A negative code is returned in error.
+ */
int snd_hdac_get_display_clk(struct hdac_bus *bus)
{
struct i915_audio_component *acomp = bus->audio_component;
@@ -134,6 +170,17 @@
return !strcmp(dev->driver->name, "i915");
}
+/**
+ * snd_hdac_i915_register_notifier - Register i915 audio component ops
+ * @aops: i915 audio component ops
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function sets the given ops to be called by the i915 graphics driver.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops)
{
if (WARN_ON(!hdac_acomp))
@@ -144,6 +191,18 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier);
+/**
+ * snd_hdac_i915_init - Initialize i915 audio component
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function initializes and sets up the audio component to communicate
+ * with i915 graphics driver.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_i915_init(struct hdac_bus *bus)
{
struct component_match *match = NULL;
@@ -187,6 +246,17 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_i915_init);
+/**
+ * snd_hdac_i915_exit - Finalize i915 audio component
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function releases the i915 audio component that has been used.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_i915_exit(struct hdac_bus *bus)
{
struct device *dev = bus->dev;
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index b0ed870..eb8f7c3 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -339,6 +339,12 @@
.use_single_rw = true,
};
+/**
+ * snd_hdac_regmap_init - Initialize regmap for HDA register accesses
+ * @codec: the codec object
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_regmap_init(struct hdac_device *codec)
{
struct regmap *regmap;
@@ -352,6 +358,10 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_regmap_init);
+/**
+ * snd_hdac_regmap_init - Release the regmap from HDA codec
+ * @codec: the codec object
+ */
void snd_hdac_regmap_exit(struct hdac_device *codec)
{
if (codec->regmap) {
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index 8981159..38990a7 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -426,7 +426,8 @@
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_setup_periods);
-/* snd_hdac_stream_set_params - set stream parameters
+/**
+ * snd_hdac_stream_set_params - set stream parameters
* @azx_dev: HD-audio core stream for which parameters are to be set
* @format_val: format value parameter
*
diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c
index c71142d..42d61bf 100644
--- a/sound/hda/hdac_sysfs.c
+++ b/sound/hda/hdac_sysfs.c
@@ -45,6 +45,13 @@
CODEC_ATTR_STR(vendor_name);
CODEC_ATTR_STR(chip_name);
+static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
+ char *buf)
+{
+ return snd_hdac_codec_modalias(dev_to_hdac_dev(dev), buf, 256);
+}
+static DEVICE_ATTR_RO(modalias);
+
static struct attribute *hdac_dev_attrs[] = {
&dev_attr_type.attr,
&dev_attr_vendor_id.attr,
@@ -54,6 +61,7 @@
&dev_attr_mfg.attr,
&dev_attr_vendor_name.attr,
&dev_attr_chip_name.attr,
+ &dev_attr_modalias.attr,
NULL
};
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 2a9f4a3..2706f27 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1864,7 +1864,7 @@
/* global setup */
pcm->info_flags = 0;
strcpy(pcm->name, "CS46xx - IEC958");
- chip->pcm_rear = pcm;
+ chip->pcm_iec958 = pcm;
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
@@ -2528,7 +2528,7 @@
#ifdef CONFIG_SND_CS46XX_NEW_DSP
if (chip->nr_ac97_codecs == 1) {
unsigned int id2 = chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]->id & 0xffff;
- if (id2 == 0x592b || id2 == 0x592d) {
+ if ((id2 & 0xfff0) == 0x5920) { /* CS4294 and CS4298 */
err = snd_ctl_add(card, snd_ctl_new1(&snd_cs46xx_front_dup_ctl, chip));
if (err < 0)
return err;
@@ -3780,6 +3780,11 @@
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
chip->in_suspend = 1;
snd_pcm_suspend_all(chip->pcm);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_pcm_suspend_all(chip->pcm_rear);
+ snd_pcm_suspend_all(chip->pcm_center_lfe);
+ snd_pcm_suspend_all(chip->pcm_iec958);
+#endif
// chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL);
// chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE);
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index d5ac25c..70671ad 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -15,21 +15,22 @@
#include "hda_local.h"
/*
- * find a matching codec preset
+ * find a matching codec id
*/
static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv)
{
struct hda_codec *codec = container_of(dev, struct hda_codec, core);
struct hda_codec_driver *driver =
container_of(drv, struct hda_codec_driver, core);
- const struct hda_codec_preset *preset;
+ const struct hda_device_id *list;
/* check probe_id instead of vendor_id if set */
u32 id = codec->probe_id ? codec->probe_id : codec->core.vendor_id;
+ u32 rev_id = codec->core.revision_id;
- for (preset = driver->preset; preset->id; preset++) {
- if (preset->id == id &&
- (!preset->rev || preset->rev == codec->core.revision_id)) {
- codec->preset = preset;
+ for (list = driver->id; list->vendor_id; list++) {
+ if (list->vendor_id == id &&
+ (!list->rev_id || list->rev_id == rev_id)) {
+ codec->preset = list;
return 1;
}
}
@@ -45,26 +46,45 @@
codec->patch_ops.unsol_event(codec, ev);
}
-/* reset the codec name from the preset */
-static int codec_refresh_name(struct hda_codec *codec, const char *name)
+/**
+ * snd_hda_codec_set_name - set the codec name
+ * @codec: the HDA codec
+ * @name: name string to set
+ */
+int snd_hda_codec_set_name(struct hda_codec *codec, const char *name)
{
- if (name) {
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup(name, GFP_KERNEL);
+ int err;
+
+ if (!name)
+ return 0;
+ err = snd_hdac_device_set_chip_name(&codec->core, name);
+ if (err < 0)
+ return err;
+
+ /* update the mixer name */
+ if (!*codec->card->mixername ||
+ codec->bus->mixer_assigned >= codec->core.addr) {
+ snprintf(codec->card->mixername,
+ sizeof(codec->card->mixername), "%s %s",
+ codec->core.vendor_name, codec->core.chip_name);
+ codec->bus->mixer_assigned = codec->core.addr;
}
- return codec->core.chip_name ? 0 : -ENOMEM;
+
+ return 0;
}
+EXPORT_SYMBOL_GPL(snd_hda_codec_set_name);
static int hda_codec_driver_probe(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
struct module *owner = dev->driver->owner;
+ hda_codec_patch_t patch;
int err;
if (WARN_ON(!codec->preset))
return -EINVAL;
- err = codec_refresh_name(codec, codec->preset->name);
+ err = snd_hda_codec_set_name(codec, codec->preset->name);
if (err < 0)
goto error;
err = snd_hdac_regmap_init(&codec->core);
@@ -76,9 +96,12 @@
goto error;
}
- err = codec->preset->patch(codec);
- if (err < 0)
- goto error_module;
+ patch = (hda_codec_patch_t)codec->preset->driver_data;
+ if (patch) {
+ err = patch(codec);
+ if (err < 0)
+ goto error_module;
+ }
err = snd_hda_codec_build_pcms(codec);
if (err < 0)
@@ -155,11 +178,10 @@
static void codec_bind_module(struct hda_codec *codec)
{
#ifdef MODULE
- request_module("snd-hda-codec-id:%08x", codec->core.vendor_id);
- if (codec_probed(codec))
- return;
- request_module("snd-hda-codec-id:%04x*",
- (codec->core.vendor_id >> 16) & 0xffff);
+ char modalias[32];
+
+ snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias));
+ request_module(modalias);
if (codec_probed(codec))
return;
#endif
@@ -251,11 +273,6 @@
}
}
- /* audio codec should override the mixer name */
- if (codec->core.afg || !*codec->card->mixername)
- snprintf(codec->card->mixername,
- sizeof(codec->card->mixername), "%s %s",
- codec->core.vendor_name, codec->core.chip_name);
return 0;
error:
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a249d54..8374188 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -91,50 +91,6 @@
}
/**
- * snd_hda_codec_read - send a command and get the response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command and read the corresponding response.
- *
- * Returns the obtained response value, or -1 for an error.
- */
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
- int flags,
- unsigned int verb, unsigned int parm)
-{
- unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
- unsigned int res;
- if (snd_hdac_exec_verb(&codec->core, cmd, flags, &res))
- return -1;
- return res;
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_read);
-
-/**
- * snd_hda_codec_write - send a single command without waiting for response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command without waiting for response.
- *
- * Returns 0 if successful, or a negative error code.
- */
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
- unsigned int verb, unsigned int parm)
-{
- unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
- return snd_hdac_exec_verb(&codec->core, cmd, flags, NULL);
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_write);
-
-/**
* snd_hda_sequence_write - sequence writes
* @codec: the HDA codec
* @seq: VERB array to send
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2970413..373fcad 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -22,6 +22,7 @@
#define __SOUND_HDA_CODEC_H
#include <linux/kref.h>
+#include <linux/mod_devicetable.h>
#include <sound/info.h>
#include <sound/control.h>
#include <sound/pcm.h>
@@ -69,6 +70,7 @@
unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
int primary_dig_out_type; /* primary digital out PCM type */
+ unsigned int mixer_assigned; /* codec addr for mixer name */
};
/* from hdac_bus to hda_bus */
@@ -80,19 +82,21 @@
* Known codecs have the patch to build and set up the controls/PCMs
* better than the generic parser.
*/
-struct hda_codec_preset {
- unsigned int id;
- unsigned int rev;
- const char *name;
- int (*patch)(struct hda_codec *codec);
-};
+typedef int (*hda_codec_patch_t)(struct hda_codec *);
#define HDA_CODEC_ID_GENERIC_HDMI 0x00000101
#define HDA_CODEC_ID_GENERIC 0x00000201
+#define HDA_CODEC_REV_ENTRY(_vid, _rev, _name, _patch) \
+ { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \
+ .api_version = HDA_DEV_LEGACY, \
+ .driver_data = (unsigned long)(_patch) }
+#define HDA_CODEC_ENTRY(_vid, _name, _patch) \
+ HDA_CODEC_REV_ENTRY(_vid, 0, _name, _patch)
+
struct hda_codec_driver {
struct hdac_driver core;
- const struct hda_codec_preset *preset;
+ const struct hda_device_id *id;
};
int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name,
@@ -183,7 +187,7 @@
u32 probe_id; /* overridden id for probing */
/* detected preset */
- const struct hda_codec_preset *preset;
+ const struct hda_device_id *preset;
const char *modelname; /* model name for preset */
/* set by patch */
@@ -297,10 +301,6 @@
/*
* constructors
*/
-int snd_hda_bus_new(struct snd_card *card,
- const struct hdac_bus_ops *ops,
- const struct hdac_io_ops *io_ops,
- struct hda_bus **busp);
int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
unsigned int codec_addr, struct hda_codec **codecp);
int snd_hda_codec_configure(struct hda_codec *codec);
@@ -309,11 +309,21 @@
/*
* low level functions
*/
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
+static inline unsigned int
+snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
int flags,
- unsigned int verb, unsigned int parm);
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
- unsigned int verb, unsigned int parm);
+ unsigned int verb, unsigned int parm)
+{
+ return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm);
+}
+
+static inline int
+snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
+ unsigned int verb, unsigned int parm)
+{
+ return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm);
+}
+
#define snd_hda_param_read(codec, nid, param) \
snd_hdac_read_parm(&(codec)->core, nid, param)
#define snd_hda_get_sub_nodes(codec, nid, start_nid) \
@@ -453,6 +463,8 @@
void snd_hda_bus_reset(struct hda_bus *bus);
void snd_hda_bus_reset_codecs(struct hda_bus *bus);
+int snd_hda_codec_set_name(struct hda_codec *codec, const char *name);
+
/*
* power management
*/
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 9444559..22dbfa5 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1045,6 +1045,7 @@
mutex_init(&bus->prepare_mutex);
bus->pci = chip->pci;
bus->modelname = model;
+ bus->mixer_assigned = -1;
bus->core.snoop = azx_snoop(chip);
if (chip->get_position[0] != azx_get_pos_lpib ||
chip->get_position[1] != azx_get_pos_lpib)
@@ -1059,6 +1060,9 @@
bus->needs_damn_long_delay = 1;
}
+ if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY)
+ bus->core.align_bdle_4k = true;
+
/* AMD chipsets often cause the communication stalls upon certain
* sequence like the pin-detection. It seems that forcing the synced
* access works around the stall. Grrr...
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 24f9111..c6e8a65 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5877,13 +5877,14 @@
return err;
}
-static const struct hda_codec_preset snd_hda_preset_generic[] = {
- { .id = HDA_CODEC_ID_GENERIC, .patch = snd_hda_parse_generic_codec },
+static const struct hda_device_id snd_hda_id_generic[] = {
+ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC, "Generic", snd_hda_parse_generic_codec),
{} /* terminator */
};
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_generic);
static struct hda_codec_driver generic_driver = {
- .preset = snd_hda_preset_generic,
+ .id = snd_hda_id_generic,
};
module_hda_codec_driver(generic_driver);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c38c68f5..4d2cbe2 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -334,6 +334,7 @@
#define AZX_DCAPS_PRESET_CTHDA \
(AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB |\
+ AZX_DCAPS_NO_64BIT |\
AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF)
/*
@@ -2104,6 +2105,11 @@
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
{ PCI_DEVICE(0x8086, 0x8d21),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+ /* Lewisburg */
+ { PCI_DEVICE(0x8086, 0xa1f0),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+ { PCI_DEVICE(0x8086, 0xa270),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
/* Lynx Point-LP */
{ PCI_DEVICE(0x8086, 0x9c20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
@@ -2284,11 +2290,13 @@
.class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
.class_mask = 0xffffff,
.driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND |
+ AZX_DCAPS_NO_64BIT |
AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB },
#else
/* this entry seems still valid -- i.e. without emu20kx chip */
{ PCI_DEVICE(0x1102, 0x0009),
.driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND |
+ AZX_DCAPS_NO_64BIT |
AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB },
#endif
/* CM8888 */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4a21c21..d0e066e 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -681,12 +681,7 @@
snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid,
unsigned int target_state)
{
- unsigned int state = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_POWER_STATE, 0);
- if (state & AC_PWRST_ERROR)
- return true;
- state = (state >> 4) & 0x0f;
- return (state == target_state);
+ return snd_hdac_check_power_state(&codec->core, nid, target_state);
}
unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index a6e3d9b..64e0d1d 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -595,8 +595,7 @@
static void parse_chip_name_mode(char *buf, struct hda_bus *bus,
struct hda_codec **codecp)
{
- kfree((*codecp)->core.chip_name);
- (*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL);
+ snd_hda_codec_set_name(*codecp, buf);
}
#define DEFINE_PARSE_ID_MODE(name) \
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index c033a4e..e0fb8c6 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1165,32 +1165,31 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_analog[] = {
- { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
- { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
- { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
- { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
- { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 },
- { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 },
- { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
- { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
- { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 },
- { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
- { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
- { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
- { .id = 0x11d4882a, .name = "AD1882A", .patch = patch_ad1882 },
- { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 },
- { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
+static const struct hda_device_id snd_hda_id_analog[] = {
+ HDA_CODEC_ENTRY(0x11d4184a, "AD1884A", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41882, "AD1882", patch_ad1882),
+ HDA_CODEC_ENTRY(0x11d41883, "AD1883", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41884, "AD1884", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d4194a, "AD1984A", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d4194b, "AD1984B", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41981, "AD1981", patch_ad1981),
+ HDA_CODEC_ENTRY(0x11d41983, "AD1983", patch_ad1983),
+ HDA_CODEC_ENTRY(0x11d41984, "AD1984", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41986, "AD1986A", patch_ad1986a),
+ HDA_CODEC_ENTRY(0x11d41988, "AD1988", patch_ad1988),
+ HDA_CODEC_ENTRY(0x11d4198b, "AD1988B", patch_ad1988),
+ HDA_CODEC_ENTRY(0x11d4882a, "AD1882A", patch_ad1882),
+ HDA_CODEC_ENTRY(0x11d4989a, "AD1989A", patch_ad1988),
+ HDA_CODEC_ENTRY(0x11d4989b, "AD1989B", patch_ad1988),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:11d4*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_analog);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Analog Devices HD-audio codec");
static struct hda_codec_driver analog_driver = {
- .preset = snd_hda_preset_analog,
+ .id = snd_hda_id_analog,
};
module_hda_codec_driver(analog_driver);
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 484bbf4..c2d9ee9 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -83,22 +83,19 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_ca0110[] = {
- { .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 },
- { .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 },
- { .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 },
+static const struct hda_device_id snd_hda_id_ca0110[] = {
+ HDA_CODEC_ENTRY(0x1102000a, "CA0110-IBG", patch_ca0110),
+ HDA_CODEC_ENTRY(0x1102000b, "CA0110-IBG", patch_ca0110),
+ HDA_CODEC_ENTRY(0x1102000d, "SB0880 X-Fi", patch_ca0110),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:1102000a");
-MODULE_ALIAS("snd-hda-codec-id:1102000b");
-MODULE_ALIAS("snd-hda-codec-id:1102000d");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0110);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec");
static struct hda_codec_driver ca0110_driver = {
- .preset = snd_hda_preset_ca0110,
+ .id = snd_hda_id_ca0110,
};
module_hda_codec_driver(ca0110_driver);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 186792f..f8a12ca4 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2673,13 +2673,13 @@
do {
if (dspload_is_loaded(codec)) {
- pr_info("ca0132 DOWNLOAD OK :-) DSP IS RUNNING.\n");
+ codec_info(codec, "ca0132 DSP downloaded and running\n");
return true;
}
msleep(20);
} while (time_before(jiffies, timeout));
- pr_err("ca0132 DOWNLOAD FAILED!!! DSP IS NOT RUNNING.\n");
+ codec_err(codec, "ca0132 failed to download DSP\n");
return false;
}
@@ -4375,7 +4375,7 @@
dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
- pr_err("ca0132 dspload_image failed.\n");
+ codec_err(codec, "ca0132 DSP load image failed\n");
goto exit_download;
}
@@ -4778,18 +4778,17 @@
/*
* patch entries
*/
-static struct hda_codec_preset snd_hda_preset_ca0132[] = {
- { .id = 0x11020011, .name = "CA0132", .patch = patch_ca0132 },
+static struct hda_device_id snd_hda_id_ca0132[] = {
+ HDA_CODEC_ENTRY(0x11020011, "CA0132", patch_ca0132),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:11020011");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0132);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Creative Sound Core3D codec");
static struct hda_codec_driver ca0132_driver = {
- .preset = snd_hda_preset_ca0132,
+ .id = snd_hda_id_ca0132,
};
module_hda_codec_driver(ca0132_driver);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 85813de..a12ae8a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -570,6 +570,7 @@
return NULL;
codec->spec = spec;
spec->vendor_nid = vendor_nid;
+ codec->power_save_node = 1;
snd_hda_gen_spec_init(&spec->gen);
return spec;
@@ -1200,26 +1201,21 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_cirrus[] = {
- { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
- { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
- { .id = 0x10134208, .name = "CS4208", .patch = patch_cs4208 },
- { .id = 0x10134210, .name = "CS4210", .patch = patch_cs4210 },
- { .id = 0x10134213, .name = "CS4213", .patch = patch_cs4213 },
+static const struct hda_device_id snd_hda_id_cirrus[] = {
+ HDA_CODEC_ENTRY(0x10134206, "CS4206", patch_cs420x),
+ HDA_CODEC_ENTRY(0x10134207, "CS4207", patch_cs420x),
+ HDA_CODEC_ENTRY(0x10134208, "CS4208", patch_cs4208),
+ HDA_CODEC_ENTRY(0x10134210, "CS4210", patch_cs4210),
+ HDA_CODEC_ENTRY(0x10134213, "CS4213", patch_cs4213),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:10134206");
-MODULE_ALIAS("snd-hda-codec-id:10134207");
-MODULE_ALIAS("snd-hda-codec-id:10134208");
-MODULE_ALIAS("snd-hda-codec-id:10134210");
-MODULE_ALIAS("snd-hda-codec-id:10134213");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cirrus);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");
static struct hda_codec_driver cirrus_driver = {
- .preset = snd_hda_preset_cirrus,
+ .id = snd_hda_id_cirrus,
};
module_hda_codec_driver(cirrus_driver);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index f5ed078..1b2195d 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -123,22 +123,19 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_cmedia[] = {
- { .id = 0x13f68888, .name = "CMI8888", .patch = patch_cmi8888 },
- { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
- { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
+static const struct hda_device_id snd_hda_id_cmedia[] = {
+ HDA_CODEC_ENTRY(0x13f68888, "CMI8888", patch_cmi8888),
+ HDA_CODEC_ENTRY(0x13f69880, "CMI9880", patch_cmi9880),
+ HDA_CODEC_ENTRY(0x434d4980, "CMI9880", patch_cmi9880),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:13f68888");
-MODULE_ALIAS("snd-hda-codec-id:13f69880");
-MODULE_ALIAS("snd-hda-codec-id:434d4980");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cmedia);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("C-Media HD-audio codec");
static struct hda_codec_driver cmedia_driver = {
- .preset = snd_hda_preset_cmedia,
+ .id = snd_hda_id_cmedia,
};
module_hda_codec_driver(cmedia_driver);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 2f0ec7c..c8b8ef5 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -954,100 +954,44 @@
/*
*/
-static const struct hda_codec_preset snd_hda_preset_conexant[] = {
- { .id = 0x14f15045, .name = "CX20549 (Venice)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15047, .name = "CX20551 (Waikiki)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15051, .name = "CX20561 (Hermosa)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15066, .name = "CX20582 (Pebble)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15068, .name = "CX20584",
- .patch = patch_conexant_auto },
- { .id = 0x14f15069, .name = "CX20585",
- .patch = patch_conexant_auto },
- { .id = 0x14f1506c, .name = "CX20588",
- .patch = patch_conexant_auto },
- { .id = 0x14f1506e, .name = "CX20590",
- .patch = patch_conexant_auto },
- { .id = 0x14f15097, .name = "CX20631",
- .patch = patch_conexant_auto },
- { .id = 0x14f15098, .name = "CX20632",
- .patch = patch_conexant_auto },
- { .id = 0x14f150a1, .name = "CX20641",
- .patch = patch_conexant_auto },
- { .id = 0x14f150a2, .name = "CX20642",
- .patch = patch_conexant_auto },
- { .id = 0x14f150ab, .name = "CX20651",
- .patch = patch_conexant_auto },
- { .id = 0x14f150ac, .name = "CX20652",
- .patch = patch_conexant_auto },
- { .id = 0x14f150b8, .name = "CX20664",
- .patch = patch_conexant_auto },
- { .id = 0x14f150b9, .name = "CX20665",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f1, .name = "CX20721",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f2, .name = "CX20722",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f3, .name = "CX20723",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f4, .name = "CX20724",
- .patch = patch_conexant_auto },
- { .id = 0x14f1510f, .name = "CX20751/2",
- .patch = patch_conexant_auto },
- { .id = 0x14f15110, .name = "CX20751/2",
- .patch = patch_conexant_auto },
- { .id = 0x14f15111, .name = "CX20753/4",
- .patch = patch_conexant_auto },
- { .id = 0x14f15113, .name = "CX20755",
- .patch = patch_conexant_auto },
- { .id = 0x14f15114, .name = "CX20756",
- .patch = patch_conexant_auto },
- { .id = 0x14f15115, .name = "CX20757",
- .patch = patch_conexant_auto },
- { .id = 0x14f151d7, .name = "CX20952",
- .patch = patch_conexant_auto },
+static const struct hda_device_id snd_hda_id_conexant[] = {
+ HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15066, "CX20582 (Pebble)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15067, "CX20583 (Pebble HSF)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15068, "CX20584", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15069, "CX20585", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f1506c, "CX20588", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f1506e, "CX20590", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15097, "CX20631", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15098, "CX20632", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150a1, "CX20641", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150a2, "CX20642", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150ab, "CX20651", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15111, "CX20753/4", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15113, "CX20755", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15114, "CX20756", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15115, "CX20757", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f151d7, "CX20952", patch_conexant_auto),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:14f15045");
-MODULE_ALIAS("snd-hda-codec-id:14f15047");
-MODULE_ALIAS("snd-hda-codec-id:14f15051");
-MODULE_ALIAS("snd-hda-codec-id:14f15066");
-MODULE_ALIAS("snd-hda-codec-id:14f15067");
-MODULE_ALIAS("snd-hda-codec-id:14f15068");
-MODULE_ALIAS("snd-hda-codec-id:14f15069");
-MODULE_ALIAS("snd-hda-codec-id:14f1506c");
-MODULE_ALIAS("snd-hda-codec-id:14f1506e");
-MODULE_ALIAS("snd-hda-codec-id:14f15097");
-MODULE_ALIAS("snd-hda-codec-id:14f15098");
-MODULE_ALIAS("snd-hda-codec-id:14f150a1");
-MODULE_ALIAS("snd-hda-codec-id:14f150a2");
-MODULE_ALIAS("snd-hda-codec-id:14f150ab");
-MODULE_ALIAS("snd-hda-codec-id:14f150ac");
-MODULE_ALIAS("snd-hda-codec-id:14f150b8");
-MODULE_ALIAS("snd-hda-codec-id:14f150b9");
-MODULE_ALIAS("snd-hda-codec-id:14f150f1");
-MODULE_ALIAS("snd-hda-codec-id:14f150f2");
-MODULE_ALIAS("snd-hda-codec-id:14f150f3");
-MODULE_ALIAS("snd-hda-codec-id:14f150f4");
-MODULE_ALIAS("snd-hda-codec-id:14f1510f");
-MODULE_ALIAS("snd-hda-codec-id:14f15110");
-MODULE_ALIAS("snd-hda-codec-id:14f15111");
-MODULE_ALIAS("snd-hda-codec-id:14f15113");
-MODULE_ALIAS("snd-hda-codec-id:14f15114");
-MODULE_ALIAS("snd-hda-codec-id:14f15115");
-MODULE_ALIAS("snd-hda-codec-id:14f151d7");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_conexant);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
static struct hda_codec_driver conexant_driver = {
- .preset = snd_hda_preset_conexant,
+ .id = snd_hda_id_conexant,
};
module_hda_codec_driver(conexant_driver);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index acbfbe08..f503a88 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1775,6 +1775,16 @@
return non_pcm;
}
+/* There is a fixed mapping between audio pin node and display port
+ * on current Intel platforms:
+ * Pin Widget 5 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 6 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 7 - PORT D (port = 3 in i915 driver)
+ */
+static int intel_pin2port(hda_nid_t pin_nid)
+{
+ return pin_nid - 4;
+}
/*
* HDMI callbacks
@@ -1791,6 +1801,8 @@
int pin_idx = hinfo_to_pin_index(codec, hinfo);
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct i915_audio_component *acomp = codec->bus->core.audio_component;
bool non_pcm;
int pinctl;
@@ -1807,6 +1819,13 @@
intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx);
}
+ /* Call sync_audio_rate to set the N/CTS/M manually if necessary */
+ /* Todo: add DP1.2 MST audio support later */
+ if (acomp && acomp->ops && acomp->ops->sync_audio_rate)
+ acomp->ops->sync_audio_rate(acomp->dev,
+ intel_pin2port(pin_nid),
+ runtime->rate);
+
non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
mutex_lock(&per_pin->lock);
per_pin->channels = substream->runtime->channels;
@@ -2561,7 +2580,7 @@
struct hdmi_spec *spec = codec->spec;
struct snd_pcm_hw_constraint_list *hw_constraints_channels = NULL;
- switch (codec->preset->id) {
+ switch (codec->preset->vendor_id) {
case 0x10de0002:
case 0x10de0003:
case 0x10de0005:
@@ -2879,7 +2898,7 @@
snd_pcm_alt_chmaps, 8, 0, &chmap);
if (err < 0)
return err;
- switch (codec->preset->id) {
+ switch (codec->preset->vendor_id) {
case 0x10de0002:
case 0x10de0003:
case 0x10de0005:
@@ -3487,138 +3506,77 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
-{ .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi },
-{ .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi },
-{ .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi },
-{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi },
-{ .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0003, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi },
-{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi },
+static const struct hda_device_id snd_hda_id_hdmi[] = {
+HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x1002aa01, "R6xx HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x10951390, "SiI1390 HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x10951392, "SiI1392 HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x17e80047, "Chrontel HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x10de0002, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0003, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0005, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0006, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0007, "MCP79/7A HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de000a, "GPU 0a HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000b, "GPU 0b HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000c, "MCP89 HDMI", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000d, "GPU 0d HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0010, "GPU 10 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0011, "GPU 11 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0012, "GPU 12 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0013, "GPU 13 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0014, "GPU 14 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0015, "GPU 15 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0016, "GPU 16 HDMI/DP", patch_nvhdmi),
/* 17 is known to be absent */
-{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0020, .name = "Tegra30 HDMI", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0022, .name = "Tegra114 HDMI", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0028, .name = "Tegra124 HDMI", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0029, .name = "Tegra210 HDMI/DP", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
-{ .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de007d, .name = "GPU 7d HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch },
-{ .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
-{ .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
-{ .id = 0x11069f84, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi },
-{ .id = 0x11069f85, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi },
-{ .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862801, .name = "Bearlake HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862802, .name = "Cantiga HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862803, .name = "Eaglelake HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862809, .name = "Skylake HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x8086280a, .name = "Broxton HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi },
+HDA_CODEC_ENTRY(0x10de0018, "GPU 18 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0019, "GPU 19 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001a, "GPU 1a HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001b, "GPU 1b HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001c, "GPU 1c HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0020, "Tegra30 HDMI", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0022, "Tegra114 HDMI", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0028, "Tegra124 HDMI", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0029, "Tegra210 HDMI/DP", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0040, "GPU 40 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0041, "GPU 41 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0042, "GPU 42 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0043, "GPU 43 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0044, "GPU 44 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0051, "GPU 51 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0060, "GPU 60 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0067, "MCP67 HDMI", patch_nvhdmi_2ch),
+HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch),
+HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi),
+HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi),
+HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi),
/* special ID for generic HDMI */
-{ .id = HDA_CODEC_ID_GENERIC_HDMI, .patch = patch_generic_hdmi },
+HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:1002793c");
-MODULE_ALIAS("snd-hda-codec-id:10027919");
-MODULE_ALIAS("snd-hda-codec-id:1002791a");
-MODULE_ALIAS("snd-hda-codec-id:1002aa01");
-MODULE_ALIAS("snd-hda-codec-id:10951390");
-MODULE_ALIAS("snd-hda-codec-id:10951392");
-MODULE_ALIAS("snd-hda-codec-id:10de0002");
-MODULE_ALIAS("snd-hda-codec-id:10de0003");
-MODULE_ALIAS("snd-hda-codec-id:10de0005");
-MODULE_ALIAS("snd-hda-codec-id:10de0006");
-MODULE_ALIAS("snd-hda-codec-id:10de0007");
-MODULE_ALIAS("snd-hda-codec-id:10de000a");
-MODULE_ALIAS("snd-hda-codec-id:10de000b");
-MODULE_ALIAS("snd-hda-codec-id:10de000c");
-MODULE_ALIAS("snd-hda-codec-id:10de000d");
-MODULE_ALIAS("snd-hda-codec-id:10de0010");
-MODULE_ALIAS("snd-hda-codec-id:10de0011");
-MODULE_ALIAS("snd-hda-codec-id:10de0012");
-MODULE_ALIAS("snd-hda-codec-id:10de0013");
-MODULE_ALIAS("snd-hda-codec-id:10de0014");
-MODULE_ALIAS("snd-hda-codec-id:10de0015");
-MODULE_ALIAS("snd-hda-codec-id:10de0016");
-MODULE_ALIAS("snd-hda-codec-id:10de0018");
-MODULE_ALIAS("snd-hda-codec-id:10de0019");
-MODULE_ALIAS("snd-hda-codec-id:10de001a");
-MODULE_ALIAS("snd-hda-codec-id:10de001b");
-MODULE_ALIAS("snd-hda-codec-id:10de001c");
-MODULE_ALIAS("snd-hda-codec-id:10de0028");
-MODULE_ALIAS("snd-hda-codec-id:10de0040");
-MODULE_ALIAS("snd-hda-codec-id:10de0041");
-MODULE_ALIAS("snd-hda-codec-id:10de0042");
-MODULE_ALIAS("snd-hda-codec-id:10de0043");
-MODULE_ALIAS("snd-hda-codec-id:10de0044");
-MODULE_ALIAS("snd-hda-codec-id:10de0051");
-MODULE_ALIAS("snd-hda-codec-id:10de0060");
-MODULE_ALIAS("snd-hda-codec-id:10de0067");
-MODULE_ALIAS("snd-hda-codec-id:10de0070");
-MODULE_ALIAS("snd-hda-codec-id:10de0071");
-MODULE_ALIAS("snd-hda-codec-id:10de0072");
-MODULE_ALIAS("snd-hda-codec-id:10de007d");
-MODULE_ALIAS("snd-hda-codec-id:10de8001");
-MODULE_ALIAS("snd-hda-codec-id:11069f80");
-MODULE_ALIAS("snd-hda-codec-id:11069f81");
-MODULE_ALIAS("snd-hda-codec-id:11069f84");
-MODULE_ALIAS("snd-hda-codec-id:11069f85");
-MODULE_ALIAS("snd-hda-codec-id:17e80047");
-MODULE_ALIAS("snd-hda-codec-id:80860054");
-MODULE_ALIAS("snd-hda-codec-id:80862801");
-MODULE_ALIAS("snd-hda-codec-id:80862802");
-MODULE_ALIAS("snd-hda-codec-id:80862803");
-MODULE_ALIAS("snd-hda-codec-id:80862804");
-MODULE_ALIAS("snd-hda-codec-id:80862805");
-MODULE_ALIAS("snd-hda-codec-id:80862806");
-MODULE_ALIAS("snd-hda-codec-id:80862807");
-MODULE_ALIAS("snd-hda-codec-id:80862808");
-MODULE_ALIAS("snd-hda-codec-id:80862809");
-MODULE_ALIAS("snd-hda-codec-id:8086280a");
-MODULE_ALIAS("snd-hda-codec-id:80862880");
-MODULE_ALIAS("snd-hda-codec-id:80862882");
-MODULE_ALIAS("snd-hda-codec-id:80862883");
-MODULE_ALIAS("snd-hda-codec-id:808629fb");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_hdmi);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("HDMI HD-audio codec");
@@ -3627,7 +3585,7 @@
MODULE_ALIAS("snd-hda-codec-atihdmi");
static struct hda_codec_driver hdmi_driver = {
- .preset = snd_hda_preset_hdmi,
+ .id = snd_hda_id_hdmi,
};
module_hda_codec_driver(hdmi_driver);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 16b8dcb..2f7b065 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -822,17 +822,7 @@
};
-/* replace the codec chip_name with the given string */
-static int alc_codec_rename(struct hda_codec *codec, const char *name)
-{
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup(name, GFP_KERNEL);
- if (!codec->core.chip_name) {
- alc_free(codec);
- return -ENOMEM;
- }
- return 0;
-}
+#define alc_codec_rename(codec, name) snd_hda_codec_set_name(codec, name)
/*
* Rename codecs appropriately from COEF value or subvendor id
@@ -4596,6 +4586,7 @@
ALC292_FIXUP_DELL_E7X,
ALC292_FIXUP_DISABLE_AAMIX,
ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC275_FIXUP_DELL_XPS,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5165,6 +5156,17 @@
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE
},
+ [ALC275_FIXUP_DELL_XPS] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Enables internal speaker */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x1f},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x00c0},
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x30},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x00b1},
+ {}
+ }
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5179,6 +5181,7 @@
SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+ SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X),
SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X),
SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER),
@@ -6627,78 +6630,70 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_realtek[] = {
- { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
- { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 },
- { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
- { .id = 0x10ec0235, .name = "ALC233", .patch = patch_alc269 },
- { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 },
- { .id = 0x10ec0256, .name = "ALC256", .patch = patch_alc269 },
- { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
- { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
- { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
- { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
- { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
- { .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 },
- { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
- { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
- { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
- { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
- { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
- { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
- { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
- { .id = 0x10ec0285, .name = "ALC285", .patch = patch_alc269 },
- { .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 },
- { .id = 0x10ec0288, .name = "ALC288", .patch = patch_alc269 },
- { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
- { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
- { .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 },
- { .id = 0x10ec0298, .name = "ALC298", .patch = patch_alc269 },
- { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
- .patch = patch_alc861 },
- { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
- { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
- { .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd },
- { .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2",
- .patch = patch_alc882 },
- { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
- .patch = patch_alc662 },
- { .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3",
- .patch = patch_alc662 },
- { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
- { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
- { .id = 0x10ec0667, .name = "ALC667", .patch = patch_alc662 },
- { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
- { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
- { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
- { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
- { .id = 0x10ec0867, .name = "ALC891", .patch = patch_alc882 },
- { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
- { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
- { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
- { .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
- .patch = patch_alc882 },
- { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
- .patch = patch_alc882 },
- { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
- { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
- { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
- .patch = patch_alc882 },
- { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
- { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
- { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
- { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 },
- { .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 },
+static const struct hda_device_id snd_hda_id_realtek[] = {
+ HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260),
+ HDA_CODEC_ENTRY(0x10ec0262, "ALC262", patch_alc262),
+ HDA_CODEC_ENTRY(0x10ec0267, "ALC267", patch_alc268),
+ HDA_CODEC_ENTRY(0x10ec0268, "ALC268", patch_alc268),
+ HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0282, "ALC282", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0283, "ALC283", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
+ HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
+ HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
+ HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
+ HDA_CODEC_ENTRY(0x10ec0862, "ALC861-VD", patch_alc861vd),
+ HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100002, "ALC662 rev2", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100101, "ALC662 rev1", patch_alc662),
+ HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100300, "ALC662 rev3", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0663, "ALC663", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0665, "ALC665", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0667, "ALC667", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0668, "ALC668", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0670, "ALC670", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0671, "ALC671", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0680, "ALC680", patch_alc680),
+ HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
+ HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0883, "ALC883", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100101, "ALC889A", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100103, "ALC889A", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0885, "ALC885", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0887, "ALC887", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0888, 0x100101, "ALC1200", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:10ec*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Realtek HD-audio codec");
static struct hda_codec_driver realtek_driver = {
- .preset = snd_hda_preset_realtek,
+ .id = snd_hda_id_realtek,
};
module_hda_codec_driver(realtek_driver);
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index 5104beb..ffda38c 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -289,41 +289,30 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_si3054[] = {
- { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x11c13055, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x11c13155, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 },
+static const struct hda_device_id snd_hda_id_si3054[] = {
+ HDA_CODEC_ENTRY(0x163c3055, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x163c3155, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x11c13026, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x11c13055, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x11c13155, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x10573055, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x10573057, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x10573155, "Si3054", patch_si3054),
/* VIA HDA on Clevo m540 */
- { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 },
+ HDA_CODEC_ENTRY(0x11063288, "Si3054", patch_si3054),
/* Asus A8J Modem (SM56) */
- { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 },
+ HDA_CODEC_ENTRY(0x15433155, "Si3054", patch_si3054),
/* LG LW20 modem */
- { .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 },
+ HDA_CODEC_ENTRY(0x18540018, "Si3054", patch_si3054),
{}
};
-
-MODULE_ALIAS("snd-hda-codec-id:163c3055");
-MODULE_ALIAS("snd-hda-codec-id:163c3155");
-MODULE_ALIAS("snd-hda-codec-id:11c13026");
-MODULE_ALIAS("snd-hda-codec-id:11c13055");
-MODULE_ALIAS("snd-hda-codec-id:11c13155");
-MODULE_ALIAS("snd-hda-codec-id:10573055");
-MODULE_ALIAS("snd-hda-codec-id:10573057");
-MODULE_ALIAS("snd-hda-codec-id:10573155");
-MODULE_ALIAS("snd-hda-codec-id:11063288");
-MODULE_ALIAS("snd-hda-codec-id:15433155");
-MODULE_ALIAS("snd-hda-codec-id:18540018");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_si3054);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Si3054 HD-audio modem codec");
static struct hda_codec_driver si3054_driver = {
- .preset = snd_hda_preset_si3054,
+ .id = snd_hda_id_si3054,
};
module_hda_codec_driver(si3054_driver);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index def5cc8..826122d 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -702,6 +702,7 @@
static bool hp_blike_system(u32 subsystem_id)
{
switch (subsystem_id) {
+ case 0x103c1473: /* HP ProBook 6550b */
case 0x103c1520:
case 0x103c1521:
case 0x103c1523:
@@ -5012,121 +5013,119 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
- { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 },
- { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x },
- { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x },
- { .id = 0x83847880, .name = "STAC9220 A2", .patch = patch_stac922x },
- { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x },
- { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x },
- { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x },
- { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac927x },
- { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac927x },
- { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac927x },
- { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac927x },
- { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac927x },
- { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac927x },
- { .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x },
- { .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x },
- { .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x },
- { .id = 0x83847623, .name = "STAC9273D", .patch = patch_stac927x },
- { .id = 0x83847624, .name = "STAC9272X", .patch = patch_stac927x },
- { .id = 0x83847625, .name = "STAC9272D", .patch = patch_stac927x },
- { .id = 0x83847626, .name = "STAC9271X", .patch = patch_stac927x },
- { .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x },
- { .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x },
- { .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x },
- { .id = 0x83847632, .name = "STAC9202", .patch = patch_stac925x },
- { .id = 0x83847633, .name = "STAC9202D", .patch = patch_stac925x },
- { .id = 0x83847634, .name = "STAC9250", .patch = patch_stac925x },
- { .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x },
- { .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x },
- { .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x },
- { .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x },
- { .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x },
- /* The following does not take into account .id=0x83847661 when subsys =
- * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
- * currently not fully supported.
- */
- { .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 },
- { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 },
- { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 },
- { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 },
- { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 },
- { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 },
- { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 },
- { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },
- { .id = 0x838476a4, .name = "STAC9255", .patch = patch_stac9205 },
- { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 },
- { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
- { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
- { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
- { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d4, .name = "92HD83C1C5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d1, .name = "92HD87B1/3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d9, .name = "92HD87B2/4", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7667, .name = "92HD88B1", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7668, .name = "92HD88B2", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7669, .name = "92HD88B4", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
- { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
- { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
- { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx },
- { .id = 0x111d7695, .name = "92HD95", .patch = patch_stac92hd95 },
- { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e8, .name = "92HD66B1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e9, .name = "92HD66B2X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ea, .name = "92HD66B3X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76eb, .name = "92HD66C1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ec, .name = "92HD66C2X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ed, .name = "92HD66C3X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ee, .name = "92HD66B1X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ef, .name = "92HD66B2X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f0, .name = "92HD66B3X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f1, .name = "92HD66C1X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f2, .name = "92HD66C2X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f3, .name = "92HD66C3/65", .patch = patch_stac92hd83xxx},
+static const struct hda_device_id snd_hda_id_sigmatel[] = {
+ HDA_CODEC_ENTRY(0x83847690, "STAC9200", patch_stac9200),
+ HDA_CODEC_ENTRY(0x83847882, "STAC9220 A1", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847680, "STAC9221 A1", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847880, "STAC9220 A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847681, "STAC9220D/9223D A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847682, "STAC9221 A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847683, "STAC9221D A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847618, "STAC9227", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847619, "STAC9227", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847616, "STAC9228", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847617, "STAC9228", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847614, "STAC9229", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847615, "STAC9229", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847620, "STAC9274", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847621, "STAC9274D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847622, "STAC9273X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847623, "STAC9273D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847624, "STAC9272X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847625, "STAC9272D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847626, "STAC9271X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847627, "STAC9271D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847628, "STAC9274X5NH", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847629, "STAC9274D5NH", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847632, "STAC9202", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847633, "STAC9202D", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847634, "STAC9250", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847635, "STAC9250D", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847636, "STAC9251", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847637, "STAC9250D", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847645, "92HD206X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847646, "92HD206D", patch_stac927x),
+ /* The following does not take into account .id=0x83847661 when subsys =
+ * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
+ * currently not fully supported.
+ */
+ HDA_CODEC_ENTRY(0x83847661, "CXD9872RD/K", patch_stac9872),
+ HDA_CODEC_ENTRY(0x83847662, "STAC9872AK", patch_stac9872),
+ HDA_CODEC_ENTRY(0x83847664, "CXD9872AKD", patch_stac9872),
+ HDA_CODEC_ENTRY(0x83847698, "STAC9205", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a0, "STAC9205", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a1, "STAC9205D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a2, "STAC9204", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a3, "STAC9204D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a4, "STAC9255", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a5, "STAC9255D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a6, "STAC9254", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a7, "STAC9254D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x111d7603, "92HD75B3X5", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d7604, "92HD83C1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d4, "92HD83C1C5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7605, "92HD81B1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d5, "92HD81B1C5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d1, "92HD87B1/3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d9, "92HD87B2/4", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7666, "92HD88B3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7667, "92HD88B1", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7668, "92HD88B2", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7669, "92HD88B4", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7608, "92HD75B2X5", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d7674, "92HD73D1X5", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d7675, "92HD73C1X5", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d7676, "92HD73E1X5", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d7695, "92HD95", patch_stac92hd95),
+ HDA_CODEC_ENTRY(0x111d76b0, "92HD71B8X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b1, "92HD71B8X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b2, "92HD71B7X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b3, "92HD71B7X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b4, "92HD71B6X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b5, "92HD71B6X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b6, "92HD71B5X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b7, "92HD71B5X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76c0, "92HD89C3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c1, "92HD89C2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c2, "92HD89C1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c3, "92HD89B3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c4, "92HD89B2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c5, "92HD89B1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c6, "92HD89E3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c7, "92HD89E2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c8, "92HD89E1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c9, "92HD89D3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76ca, "92HD89D2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76cb, "92HD89D1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76cc, "92HD89F3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76cd, "92HD89F2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76ce, "92HD89F1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76df, "92HD93BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e0, "92HD91BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e3, "92HD98BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e5, "92HD99BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e7, "92HD90BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e8, "92HD66B1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e9, "92HD66B2X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ea, "92HD66B3X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76eb, "92HD66C1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ec, "92HD66C2X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ed, "92HD66C3X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ee, "92HD66B1X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ef, "92HD66B2X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f0, "92HD66B3X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f1, "92HD66C1X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f2, "92HD66C2X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f3, "92HD66C3/65", patch_stac92hd83xxx),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:8384*");
-MODULE_ALIAS("snd-hda-codec-id:111d*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_sigmatel);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec");
static struct hda_codec_driver sigmatel_driver = {
- .preset = snd_hda_preset_sigmatel,
+ .id = snd_hda_id_sigmatel,
};
module_hda_codec_driver(sigmatel_driver);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index da53664..fc30d1e 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -785,21 +785,11 @@
override_mic_boost(codec, 0x1e, 0, 3, 40);
/* correct names for VT1708BCE */
- if (get_codec_type(codec) == VT1708BCE) {
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL);
- snprintf(codec->card->mixername,
- sizeof(codec->card->mixername),
- "%s %s", codec->core.vendor_name, codec->core.chip_name);
- }
+ if (get_codec_type(codec) == VT1708BCE)
+ snd_hda_codec_set_name(codec, "VT1708BCE");
/* correct names for VT1705 */
- if (codec->core.vendor_id == 0x11064397) {
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL);
- snprintf(codec->card->mixername,
- sizeof(codec->card->mixername),
- "%s %s", codec->core.vendor_name, codec->core.chip_name);
- }
+ if (codec->core.vendor_id == 0x11064397)
+ snd_hda_codec_set_name(codec, "VT1705");
/* automatic parse from the BIOS config */
err = via_parse_auto_config(codec);
@@ -1210,109 +1200,64 @@
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_via[] = {
- { .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x1106e710, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e711, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e712, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e713, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e714, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e715, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e716, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e717, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e720, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e721, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e722, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e723, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e724, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e725, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e726, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e727, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x11060397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11061397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11062397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11063397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11064397, .name = "VT1705",
- .patch = patch_vt1708S},
- { .id = 0x11065397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11066397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11067397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11060398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11061398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11062398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11063398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11064398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11065398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11066398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11067398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11060428, .name = "VT1718S",
- .patch = patch_vt1718S},
- { .id = 0x11064428, .name = "VT1718S",
- .patch = patch_vt1718S},
- { .id = 0x11060441, .name = "VT2020",
- .patch = patch_vt1718S},
- { .id = 0x11064441, .name = "VT1828S",
- .patch = patch_vt1718S},
- { .id = 0x11060433, .name = "VT1716S",
- .patch = patch_vt1716S},
- { .id = 0x1106a721, .name = "VT1716S",
- .patch = patch_vt1716S},
- { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P},
- { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P},
- { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812},
- { .id = 0x11060440, .name = "VT1818S",
- .patch = patch_vt1708S},
- { .id = 0x11060446, .name = "VT1802",
- .patch = patch_vt2002P},
- { .id = 0x11068446, .name = "VT1802",
- .patch = patch_vt2002P},
- { .id = 0x11064760, .name = "VT1705CF",
- .patch = patch_vt3476},
- { .id = 0x11064761, .name = "VT1708SCE",
- .patch = patch_vt3476},
- { .id = 0x11064762, .name = "VT1808",
- .patch = patch_vt3476},
+static const struct hda_device_id snd_hda_id_via[] = {
+ HDA_CODEC_ENTRY(0x11061708, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x11061709, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x1106170a, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x1106170b, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x1106e710, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e711, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e712, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e713, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e714, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e715, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e716, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e717, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e720, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e721, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e722, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e723, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e724, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e725, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e726, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e727, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x11060397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11061397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11062397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11063397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11064397, "VT1705", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11065397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11066397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11067397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11060398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11061398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11062398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11063398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11064398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11065398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11066398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11067398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11060428, "VT1718S", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11064428, "VT1718S", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11060441, "VT2020", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11064441, "VT1828S", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11060433, "VT1716S", patch_vt1716S),
+ HDA_CODEC_ENTRY(0x1106a721, "VT1716S", patch_vt1716S),
+ HDA_CODEC_ENTRY(0x11060438, "VT2002P", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11064438, "VT2002P", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11060448, "VT1812", patch_vt1812),
+ HDA_CODEC_ENTRY(0x11060440, "VT1818S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11060446, "VT1802", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11068446, "VT1802", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11064760, "VT1705CF", patch_vt3476),
+ HDA_CODEC_ENTRY(0x11064761, "VT1708SCE", patch_vt3476),
+ HDA_CODEC_ENTRY(0x11064762, "VT1808", patch_vt3476),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:1106*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_via);
static struct hda_codec_driver via_driver = {
- .preset = snd_hda_preset_via,
+ .id = snd_hda_id_via,
};
MODULE_LICENSE("GPL");
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 7acbc21..9e1ad11 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1394,7 +1394,9 @@
spin_unlock_irqrestore(&korg1212->lock, flags);
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, kPlayBufferFrames, kPlayBufferFrames);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ kPlayBufferFrames);
+
return 0;
}
@@ -1422,8 +1424,8 @@
spin_unlock_irqrestore(&korg1212->lock, flags);
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- kPlayBufferFrames, kPlayBufferFrames);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ kPlayBufferFrames);
return 0;
}
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index cba89be..8b8e2e5 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -234,8 +234,8 @@
/* the clock rate cannot be changed */
board_rate = chip->board_sample_rate;
- err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
- board_rate, board_rate);
+ err = snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_RATE,
+ board_rate);
if (err < 0) {
dev_warn(chip->card->dev, "could not constrain periods\n");
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 72e89ce..17ae926 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -1929,15 +1929,32 @@
return;
snd_m3_outw(chip, val, CODEC_DATA);
snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+ /*
+ * Workaround for buggy ES1988 integrated AC'97 codec. It remains silent
+ * until the MASTER volume or mute is touched (alsactl restore does not
+ * work).
+ */
+ if (ac97->id == 0x45838308 && reg == AC97_MASTER) {
+ snd_m3_ac97_wait(chip);
+ snd_m3_outw(chip, val, CODEC_DATA);
+ snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+ }
}
-static void snd_m3_remote_codec_config(int io, int isremote)
+static void snd_m3_remote_codec_config(struct snd_m3 *chip, int isremote)
{
+ int io = chip->iobase;
+ u16 tmp;
+
isremote = isremote ? 1 : 0;
- outw((inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK) | isremote,
- io + RING_BUS_CTRL_B);
+ tmp = inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK;
+ /* enable dock on Dell Latitude C810 */
+ if (chip->pci->subsystem_vendor == 0x1028 &&
+ chip->pci->subsystem_device == 0x00e5)
+ tmp |= M3I_DOCK_ENABLE;
+ outw(tmp | isremote, io + RING_BUS_CTRL_B);
outw((inw(io + SDO_OUT_DEST_CTRL) & ~COMMAND_ADDR_OUT) | isremote,
io + SDO_OUT_DEST_CTRL);
outw((inw(io + SDO_IN_DEST_CTRL) & ~STATUS_ADDR_IN) | isremote,
@@ -1989,7 +2006,7 @@
if (!chip->irda_workaround)
dir |= 0x10; /* assuming pci bus master? */
- snd_m3_remote_codec_config(io, 0);
+ snd_m3_remote_codec_config(chip, 0);
outw(IO_SRAM_ENABLE, io + RING_BUS_CTRL_A);
udelay(20);
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 23d7f5d..cd94ac5 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -831,9 +831,9 @@
static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_runtime *runtime)
{
if (! rme32->fullduplex_mode) {
- snd_pcm_hw_constraint_minmax(runtime,
+ snd_pcm_hw_constraint_single(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
- RME32_BUFFER_SIZE, RME32_BUFFER_SIZE);
+ RME32_BUFFER_SIZE);
snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
&hw_constraints_period_bytes);
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 2306ccf..714df90 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -1152,13 +1152,13 @@
{
unsigned int size;
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
- RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ RME96_BUFFER_SIZE);
if ((size = rme96->playback_periodsize) != 0 ||
(size = rme96->capture_periodsize) != 0)
- snd_pcm_hw_constraint_minmax(runtime,
+ snd_pcm_hw_constraint_single(runtime,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
- size, size);
+ size);
else
snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 9bba275..2875b4f 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5112,6 +5112,7 @@
dev_err(hdsp->card->dev,
"too short firmware size %d (expected %d)\n",
(int)fw->size, HDSP_FIRMWARE_SIZE);
+ release_firmware(fw);
return -EINVAL;
}
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cb666c7..8bc8016 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6080,18 +6080,17 @@
SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
32, 4096);
/* RayDAT & AIO have a fixed buffer of 16384 samples per channel */
- snd_pcm_hw_constraint_minmax(runtime,
+ snd_pcm_hw_constraint_single(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
- 16384, 16384);
+ 16384);
break;
default:
snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
64, 8192);
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS,
- 2, 2);
+ snd_pcm_hw_constraint_single(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS, 2);
break;
}
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 225bfda..7ff7d88 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -9,7 +9,6 @@
select SND_JACK if INPUT=y || INPUT=SND
select REGMAP_I2C if I2C
select REGMAP_SPI if SPI_MASTER
- select SND_COMPRESS_OFFLOAD
---help---
If you want ASoC support, you should say Y here and also to the
@@ -30,6 +29,10 @@
bool
select SND_DMAENGINE_PCM
+config SND_SOC_COMPRESS
+ bool
+ select SND_COMPRESS_OFFLOAD
+
config SND_SOC_TOPOLOGY
bool
@@ -58,6 +61,7 @@
source "sound/soc/sirf/Kconfig"
source "sound/soc/spear/Kconfig"
source "sound/soc/sti/Kconfig"
+source "sound/soc/sunxi/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
source "sound/soc/ux500/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 134aca1..8eb06db 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,6 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
-snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o
+snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o
+snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
ifneq ($(CONFIG_SND_SOC_TOPOLOGY),)
snd-soc-core-objs += soc-topology.o
@@ -40,6 +41,7 @@
obj-$(CONFIG_SND_SOC) += sirf/
obj-$(CONFIG_SND_SOC) += spear/
obj-$(CONFIG_SND_SOC) += sti/
+obj-$(CONFIG_SND_SOC) += sunxi/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
obj-$(CONFIG_SND_SOC) += ux500/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 1489cd4..2d30464 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -59,4 +59,13 @@
help
Say Y if you want to add support for audio SoC on an
at91sam9x5 based board that is using WM8731 codec.
+
+config SND_ATMEL_SOC_CLASSD
+ tristate "Atmel ASoC driver for boards using CLASSD"
+ depends on ARCH_AT91 || COMPILE_TEST
+ select SND_ATMEL_SOC_DMA
+ select REGMAP_MMIO
+ help
+ Say Y if you want to add support for Atmel ASoC driver for boards using
+ CLASSD.
endif
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index b327e5c..f6f7db4 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -11,7 +11,9 @@
snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
snd-atmel-soc-wm8904-objs := atmel_wm8904.o
snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
+snd-atmel-soc-classd-objs := atmel-classd.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
+obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
new file mode 100644
index 0000000..8276675
--- /dev/null
+++ b/sound/soc/atmel/atmel-classd.c
@@ -0,0 +1,679 @@
+/* Atmel ALSA SoC Audio Class D Amplifier (CLASSD) driver
+ *
+ * Copyright (C) 2015 Atmel
+ *
+ * Author: Songjun Wu <songjun.wu@atmel.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 or later
+ * as published by the Free Software Foundation.
+ */
+
+#include <linux/of.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include "atmel-classd.h"
+
+struct atmel_classd_pdata {
+ bool non_overlap_enable;
+ int non_overlap_time;
+ int pwm_type;
+ const char *card_name;
+};
+
+struct atmel_classd {
+ dma_addr_t phy_base;
+ struct regmap *regmap;
+ struct clk *pclk;
+ struct clk *gclk;
+ struct clk *aclk;
+ int irq;
+ const struct atmel_classd_pdata *pdata;
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id atmel_classd_of_match[] = {
+ {
+ .compatible = "atmel,sama5d2-classd",
+ }, {
+ /* sentinel */
+ }
+};
+MODULE_DEVICE_TABLE(of, atmel_classd_of_match);
+
+static struct atmel_classd_pdata *atmel_classd_dt_init(struct device *dev)
+{
+ struct device_node *np = dev->of_node;
+ struct atmel_classd_pdata *pdata;
+ const char *pwm_type;
+ int ret;
+
+ if (!np) {
+ dev_err(dev, "device node not found\n");
+ return ERR_PTR(-EINVAL);
+ }
+
+ pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata)
+ return ERR_PTR(-ENOMEM);
+
+ ret = of_property_read_string(np, "atmel,pwm-type", &pwm_type);
+ if ((ret == 0) && (strcmp(pwm_type, "diff") == 0))
+ pdata->pwm_type = CLASSD_MR_PWMTYP_DIFF;
+ else
+ pdata->pwm_type = CLASSD_MR_PWMTYP_SINGLE;
+
+ ret = of_property_read_u32(np,
+ "atmel,non-overlap-time", &pdata->non_overlap_time);
+ if (ret)
+ pdata->non_overlap_enable = false;
+ else
+ pdata->non_overlap_enable = true;
+
+ ret = of_property_read_string(np, "atmel,model", &pdata->card_name);
+ if (ret)
+ pdata->card_name = "CLASSD";
+
+ return pdata;
+}
+#else
+static inline struct atmel_classd_pdata *
+atmel_classd_dt_init(struct device *dev)
+{
+ return ERR_PTR(-EINVAL);
+}
+#endif
+
+#define ATMEL_CLASSD_RATES (SNDRV_PCM_RATE_8000 \
+ | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 \
+ | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 \
+ | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 \
+ | SNDRV_PCM_RATE_96000)
+
+static const struct snd_pcm_hardware atmel_classd_hw = {
+ .info = SNDRV_PCM_INFO_MMAP
+ | SNDRV_PCM_INFO_MMAP_VALID
+ | SNDRV_PCM_INFO_INTERLEAVED
+ | SNDRV_PCM_INFO_RESUME
+ | SNDRV_PCM_INFO_PAUSE,
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = ATMEL_CLASSD_RATES,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 256,
+ .period_bytes_max = 32 * 1024,
+ .periods_min = 2,
+ .periods_max = 256,
+};
+
+#define ATMEL_CLASSD_PREALLOC_BUF_SIZE (64 * 1024)
+
+/* cpu dai component */
+static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ regmap_write(dd->regmap, CLASSD_THR, 0x0);
+
+ return clk_prepare_enable(dd->pclk);
+}
+
+static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ clk_disable_unprepare(dd->pclk);
+}
+
+static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = {
+ .startup = atmel_classd_cpu_dai_startup,
+ .shutdown = atmel_classd_cpu_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver atmel_classd_cpu_dai = {
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ATMEL_CLASSD_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &atmel_classd_cpu_dai_ops,
+};
+
+static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = {
+ .name = "atmel-classd",
+};
+
+/* platform */
+static int
+atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct dma_slave_config *slave_config)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ if (params_physical_width(params) != 16) {
+ dev_err(rtd->platform->dev,
+ "only supports 16-bit audio data\n");
+ return -EINVAL;
+ }
+
+ slave_config->direction = DMA_MEM_TO_DEV;
+ slave_config->dst_addr = dd->phy_base + CLASSD_THR;
+ slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ slave_config->dst_maxburst = 1;
+ slave_config->src_maxburst = 1;
+ slave_config->device_fc = false;
+
+ return 0;
+}
+
+static const struct snd_dmaengine_pcm_config
+atmel_classd_dmaengine_pcm_config = {
+ .prepare_slave_config = atmel_classd_platform_configure_dma,
+ .pcm_hardware = &atmel_classd_hw,
+ .prealloc_buffer_size = ATMEL_CLASSD_PREALLOC_BUF_SIZE,
+};
+
+/* codec */
+static const char * const mono_mode_text[] = {
+ "mix", "sat", "left", "right"
+};
+
+static SOC_ENUM_SINGLE_DECL(classd_mono_mode_enum,
+ CLASSD_INTPMR, CLASSD_INTPMR_MONO_MODE_SHIFT,
+ mono_mode_text);
+
+static const char * const eqcfg_text[] = {
+ "Treble-12dB", "Treble-6dB",
+ "Medium-8dB", "Medium-3dB",
+ "Bass-12dB", "Bass-6dB",
+ "0 dB",
+ "Bass+6dB", "Bass+12dB",
+ "Medium+3dB", "Medium+8dB",
+ "Treble+6dB", "Treble+12dB",
+};
+
+static const unsigned int eqcfg_value[] = {
+ CLASSD_INTPMR_EQCFG_T_CUT_12, CLASSD_INTPMR_EQCFG_T_CUT_6,
+ CLASSD_INTPMR_EQCFG_M_CUT_8, CLASSD_INTPMR_EQCFG_M_CUT_3,
+ CLASSD_INTPMR_EQCFG_B_CUT_12, CLASSD_INTPMR_EQCFG_B_CUT_6,
+ CLASSD_INTPMR_EQCFG_FLAT,
+ CLASSD_INTPMR_EQCFG_B_BOOST_6, CLASSD_INTPMR_EQCFG_B_BOOST_12,
+ CLASSD_INTPMR_EQCFG_M_BOOST_3, CLASSD_INTPMR_EQCFG_M_BOOST_8,
+ CLASSD_INTPMR_EQCFG_T_BOOST_6, CLASSD_INTPMR_EQCFG_T_BOOST_12,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(classd_eqcfg_enum,
+ CLASSD_INTPMR, CLASSD_INTPMR_EQCFG_SHIFT, 0xf,
+ eqcfg_text, eqcfg_value);
+
+static const DECLARE_TLV_DB_SCALE(classd_digital_tlv, -7800, 100, 1);
+
+static const struct snd_kcontrol_new atmel_classd_snd_controls[] = {
+SOC_DOUBLE_TLV("Playback Volume", CLASSD_INTPMR,
+ CLASSD_INTPMR_ATTL_SHIFT, CLASSD_INTPMR_ATTR_SHIFT,
+ 78, 1, classd_digital_tlv),
+
+SOC_SINGLE("Deemphasis Switch", CLASSD_INTPMR,
+ CLASSD_INTPMR_DEEMP_SHIFT, 1, 0),
+
+SOC_SINGLE("Mono Switch", CLASSD_INTPMR, CLASSD_INTPMR_MONO_SHIFT, 1, 0),
+
+SOC_SINGLE("Swap Switch", CLASSD_INTPMR, CLASSD_INTPMR_SWAP_SHIFT, 1, 0),
+
+SOC_ENUM("Mono Mode", classd_mono_mode_enum),
+
+SOC_ENUM("EQ", classd_eqcfg_enum),
+};
+
+static const char * const pwm_type[] = {
+ "Single ended", "Differential"
+};
+
+static int atmel_classd_codec_probe(struct snd_soc_codec *codec)
+{
+ struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec);
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(card);
+ const struct atmel_classd_pdata *pdata = dd->pdata;
+ u32 mask, val;
+
+ mask = CLASSD_MR_PWMTYP_MASK;
+ val = pdata->pwm_type << CLASSD_MR_PWMTYP_SHIFT;
+
+ mask |= CLASSD_MR_NON_OVERLAP_MASK;
+ if (pdata->non_overlap_enable) {
+ val |= (CLASSD_MR_NON_OVERLAP_EN
+ << CLASSD_MR_NON_OVERLAP_SHIFT);
+
+ mask |= CLASSD_MR_NOVR_VAL_MASK;
+ switch (pdata->non_overlap_time) {
+ case 5:
+ val |= (CLASSD_MR_NOVR_VAL_5NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ case 10:
+ val |= (CLASSD_MR_NOVR_VAL_10NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ case 15:
+ val |= (CLASSD_MR_NOVR_VAL_15NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ case 20:
+ val |= (CLASSD_MR_NOVR_VAL_20NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ default:
+ val |= (CLASSD_MR_NOVR_VAL_10NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ dev_warn(codec->dev,
+ "non-overlapping value %d is invalid, the default value 10 is specified\n",
+ pdata->non_overlap_time);
+ break;
+ }
+ }
+
+ snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+ dev_info(codec->dev,
+ "PWM modulation type is %s, non-overlapping is %s\n",
+ pwm_type[pdata->pwm_type],
+ pdata->non_overlap_enable?"enabled":"disabled");
+
+ return 0;
+}
+
+static struct regmap *atmel_classd_codec_get_remap(struct device *dev)
+{
+ return dev_get_regmap(dev, NULL);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_classd = {
+ .probe = atmel_classd_codec_probe,
+ .controls = atmel_classd_snd_controls,
+ .num_controls = ARRAY_SIZE(atmel_classd_snd_controls),
+ .get_regmap = atmel_classd_codec_get_remap,
+};
+
+/* codec dai component */
+static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+ int ret;
+
+ ret = clk_prepare_enable(dd->aclk);
+ if (ret)
+ return ret;
+
+ return clk_prepare_enable(dd->gclk);
+}
+
+static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai,
+ int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u32 mask, val;
+
+ mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK;
+
+ if (mute)
+ val = mask;
+ else
+ val = 0;
+
+ snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+ return 0;
+}
+
+#define CLASSD_ACLK_RATE_11M2896_MPY_8 (112896 * 100 * 8)
+#define CLASSD_ACLK_RATE_12M288_MPY_8 (12228 * 1000 * 8)
+
+static struct {
+ int rate;
+ int sample_rate;
+ int dsp_clk;
+ unsigned long aclk_rate;
+} const sample_rates[] = {
+ { 8000, CLASSD_INTPMR_FRAME_8K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 16000, CLASSD_INTPMR_FRAME_16K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 32000, CLASSD_INTPMR_FRAME_32K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 48000, CLASSD_INTPMR_FRAME_48K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 96000, CLASSD_INTPMR_FRAME_96K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 22050, CLASSD_INTPMR_FRAME_22K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+ { 44100, CLASSD_INTPMR_FRAME_44K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+ { 88200, CLASSD_INTPMR_FRAME_88K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+};
+
+static int
+atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int fs;
+ int i, best, best_val, cur_val, ret;
+ u32 mask, val;
+
+ fs = params_rate(params);
+
+ best = 0;
+ best_val = abs(fs - sample_rates[0].rate);
+ for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+ /* Closest match */
+ cur_val = abs(fs - sample_rates[i].rate);
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+
+ dev_dbg(codec->dev,
+ "Selected SAMPLE_RATE of %dHz, ACLK_RATE of %ldHz\n",
+ sample_rates[best].rate, sample_rates[best].aclk_rate);
+
+ clk_disable_unprepare(dd->gclk);
+ clk_disable_unprepare(dd->aclk);
+
+ ret = clk_set_rate(dd->aclk, sample_rates[best].aclk_rate);
+ if (ret)
+ return ret;
+
+ mask = CLASSD_INTPMR_DSP_CLK_FREQ_MASK | CLASSD_INTPMR_FRAME_MASK;
+ val = (sample_rates[best].dsp_clk << CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT)
+ | (sample_rates[best].sample_rate << CLASSD_INTPMR_FRAME_SHIFT);
+
+ snd_soc_update_bits(codec, CLASSD_INTPMR, mask, val);
+
+ ret = clk_prepare_enable(dd->aclk);
+ if (ret)
+ return ret;
+
+ return clk_prepare_enable(dd->gclk);
+}
+
+static void
+atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ clk_disable_unprepare(dd->gclk);
+ clk_disable_unprepare(dd->aclk);
+}
+
+static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ snd_soc_update_bits(codec, CLASSD_MR,
+ CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK,
+ (CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT)
+ |(CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT));
+
+ return 0;
+}
+
+static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u32 mask, val;
+
+ mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ val = mask;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ val = (CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT)
+ | (CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = {
+ .digital_mute = atmel_classd_codec_dai_digital_mute,
+ .startup = atmel_classd_codec_dai_startup,
+ .shutdown = atmel_classd_codec_dai_shutdown,
+ .hw_params = atmel_classd_codec_dai_hw_params,
+ .prepare = atmel_classd_codec_dai_prepare,
+ .trigger = atmel_classd_codec_dai_trigger,
+};
+
+#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi"
+
+static struct snd_soc_dai_driver atmel_classd_codec_dai = {
+ .name = ATMEL_CLASSD_CODEC_DAI_NAME,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ATMEL_CLASSD_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &atmel_classd_codec_dai_ops,
+};
+
+/* ASoC sound card */
+static int atmel_classd_asoc_card_init(struct device *dev,
+ struct snd_soc_card *card)
+{
+ struct snd_soc_dai_link *dai_link;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(card);
+
+ dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL);
+ if (!dai_link)
+ return -ENOMEM;
+
+ dai_link->name = "CLASSD";
+ dai_link->stream_name = "CLASSD PCM";
+ dai_link->codec_dai_name = ATMEL_CLASSD_CODEC_DAI_NAME;
+ dai_link->cpu_dai_name = dev_name(dev);
+ dai_link->codec_name = dev_name(dev);
+ dai_link->platform_name = dev_name(dev);
+
+ card->dai_link = dai_link;
+ card->num_links = 1;
+ card->name = dd->pdata->card_name;
+ card->dev = dev;
+
+ return 0;
+};
+
+/* regmap configuration */
+static const struct reg_default atmel_classd_reg_defaults[] = {
+ { CLASSD_INTPMR, 0x00301212 },
+};
+
+#define ATMEL_CLASSD_REG_MAX 0xE4
+static const struct regmap_config atmel_classd_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = ATMEL_CLASSD_REG_MAX,
+
+ .cache_type = REGCACHE_FLAT,
+ .reg_defaults = atmel_classd_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(atmel_classd_reg_defaults),
+};
+
+static int atmel_classd_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct atmel_classd *dd;
+ struct resource *res;
+ void __iomem *io_base;
+ const struct atmel_classd_pdata *pdata;
+ struct snd_soc_card *card;
+ int ret;
+
+ pdata = dev_get_platdata(dev);
+ if (!pdata) {
+ pdata = atmel_classd_dt_init(dev);
+ if (IS_ERR(pdata))
+ return PTR_ERR(pdata);
+ }
+
+ dd = devm_kzalloc(dev, sizeof(*dd), GFP_KERNEL);
+ if (!dd)
+ return -ENOMEM;
+
+ dd->pdata = pdata;
+
+ dd->irq = platform_get_irq(pdev, 0);
+ if (dd->irq < 0) {
+ ret = dd->irq;
+ dev_err(dev, "failed to could not get irq: %d\n", ret);
+ return ret;
+ }
+
+ dd->pclk = devm_clk_get(dev, "pclk");
+ if (IS_ERR(dd->pclk)) {
+ ret = PTR_ERR(dd->pclk);
+ dev_err(dev, "failed to get peripheral clock: %d\n", ret);
+ return ret;
+ }
+
+ dd->gclk = devm_clk_get(dev, "gclk");
+ if (IS_ERR(dd->gclk)) {
+ ret = PTR_ERR(dd->gclk);
+ dev_err(dev, "failed to get GCK clock: %d\n", ret);
+ return ret;
+ }
+
+ dd->aclk = devm_clk_get(dev, "aclk");
+ if (IS_ERR(dd->aclk)) {
+ ret = PTR_ERR(dd->aclk);
+ dev_err(dev, "failed to get audio clock: %d\n", ret);
+ return ret;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ dev_err(dev, "no memory resource\n");
+ return -ENXIO;
+ }
+
+ io_base = devm_ioremap_resource(dev, res);
+ if (IS_ERR(io_base)) {
+ ret = PTR_ERR(io_base);
+ dev_err(dev, "failed to remap register memory: %d\n", ret);
+ return ret;
+ }
+
+ dd->phy_base = res->start;
+
+ dd->regmap = devm_regmap_init_mmio(dev, io_base,
+ &atmel_classd_regmap_config);
+ if (IS_ERR(dd->regmap)) {
+ ret = PTR_ERR(dd->regmap);
+ dev_err(dev, "failed to init register map: %d\n", ret);
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_component(dev,
+ &atmel_classd_cpu_dai_component,
+ &atmel_classd_cpu_dai, 1);
+ if (ret) {
+ dev_err(dev, "could not register CPU DAI: %d\n", ret);
+ return ret;
+ }
+
+ ret = devm_snd_dmaengine_pcm_register(dev,
+ &atmel_classd_dmaengine_pcm_config,
+ 0);
+ if (ret) {
+ dev_err(dev, "could not register platform: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(dev, &soc_codec_dev_classd,
+ &atmel_classd_codec_dai, 1);
+ if (ret) {
+ dev_err(dev, "could not register codec: %d\n", ret);
+ return ret;
+ }
+
+ /* register sound card */
+ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
+ snd_soc_card_set_drvdata(card, dd);
+ platform_set_drvdata(pdev, card);
+
+ ret = atmel_classd_asoc_card_init(dev, card);
+ if (ret) {
+ dev_err(dev, "failed to init sound card\n");
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_card(dev, card);
+ if (ret) {
+ dev_err(dev, "failed to register sound card: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int atmel_classd_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver atmel_classd_driver = {
+ .driver = {
+ .name = "atmel-classd",
+ .of_match_table = of_match_ptr(atmel_classd_of_match),
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = atmel_classd_probe,
+ .remove = atmel_classd_remove,
+};
+module_platform_driver(atmel_classd_driver);
+
+MODULE_DESCRIPTION("Atmel ClassD driver under ALSA SoC architecture");
+MODULE_AUTHOR("Songjun Wu <songjun.wu@atmel.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel-classd.h b/sound/soc/atmel/atmel-classd.h
new file mode 100644
index 0000000..73f8fdd
--- /dev/null
+++ b/sound/soc/atmel/atmel-classd.h
@@ -0,0 +1,120 @@
+#ifndef __ATMEL_CLASSD_H_
+#define __ATMEL_CLASSD_H_
+
+#define CLASSD_CR 0x00000000
+#define CLASSD_CR_RESET 0x1
+
+#define CLASSD_MR 0x00000004
+
+#define CLASSD_MR_LEN_DIS 0x0
+#define CLASSD_MR_LEN_EN 0x1
+#define CLASSD_MR_LEN_MASK (0x1 << 0)
+#define CLASSD_MR_LEN_SHIFT (0)
+
+#define CLASSD_MR_LMUTE_DIS 0x0
+#define CLASSD_MR_LMUTE_EN 0x1
+#define CLASSD_MR_LMUTE_SHIFT (0x1)
+#define CLASSD_MR_LMUTE_MASK (0x1 << 1)
+
+#define CLASSD_MR_REN_DIS 0x0
+#define CLASSD_MR_REN_EN 0x1
+#define CLASSD_MR_REN_MASK (0x1 << 4)
+#define CLASSD_MR_REN_SHIFT (4)
+
+#define CLASSD_MR_RMUTE_DIS 0x0
+#define CLASSD_MR_RMUTE_EN 0x1
+#define CLASSD_MR_RMUTE_SHIFT (0x5)
+#define CLASSD_MR_RMUTE_MASK (0x1 << 5)
+
+#define CLASSD_MR_PWMTYP_SINGLE 0x0
+#define CLASSD_MR_PWMTYP_DIFF 0x1
+#define CLASSD_MR_PWMTYP_MASK (0x1 << 8)
+#define CLASSD_MR_PWMTYP_SHIFT (8)
+
+#define CLASSD_MR_NON_OVERLAP_DIS 0x0
+#define CLASSD_MR_NON_OVERLAP_EN 0x1
+#define CLASSD_MR_NON_OVERLAP_MASK (0x1 << 16)
+#define CLASSD_MR_NON_OVERLAP_SHIFT (16)
+
+#define CLASSD_MR_NOVR_VAL_5NS 0x0
+#define CLASSD_MR_NOVR_VAL_10NS 0x1
+#define CLASSD_MR_NOVR_VAL_15NS 0x2
+#define CLASSD_MR_NOVR_VAL_20NS 0x3
+#define CLASSD_MR_NOVR_VAL_MASK (0x3 << 20)
+#define CLASSD_MR_NOVR_VAL_SHIFT (20)
+
+#define CLASSD_INTPMR 0x00000008
+
+#define CLASSD_INTPMR_ATTL_MASK (0x3f << 0)
+#define CLASSD_INTPMR_ATTL_SHIFT (0)
+#define CLASSD_INTPMR_ATTR_MASK (0x3f << 8)
+#define CLASSD_INTPMR_ATTR_SHIFT (8)
+
+#define CLASSD_INTPMR_DSP_CLK_FREQ_12M288 0x0
+#define CLASSD_INTPMR_DSP_CLK_FREQ_11M2896 0x1
+#define CLASSD_INTPMR_DSP_CLK_FREQ_MASK (0x1 << 16)
+#define CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT (16)
+
+#define CLASSD_INTPMR_DEEMP_DIS 0x0
+#define CLASSD_INTPMR_DEEMP_EN 0x1
+#define CLASSD_INTPMR_DEEMP_MASK (0x1 << 18)
+#define CLASSD_INTPMR_DEEMP_SHIFT (18)
+
+#define CLASSD_INTPMR_SWAP_LEFT_ON_LSB 0x0
+#define CLASSD_INTPMR_SWAP_RIGHT_ON_LSB 0x1
+#define CLASSD_INTPMR_SWAP_MASK (0x1 << 19)
+#define CLASSD_INTPMR_SWAP_SHIFT (19)
+
+#define CLASSD_INTPMR_FRAME_8K 0x0
+#define CLASSD_INTPMR_FRAME_16K 0x1
+#define CLASSD_INTPMR_FRAME_32K 0x2
+#define CLASSD_INTPMR_FRAME_48K 0x3
+#define CLASSD_INTPMR_FRAME_96K 0x4
+#define CLASSD_INTPMR_FRAME_22K 0x5
+#define CLASSD_INTPMR_FRAME_44K 0x6
+#define CLASSD_INTPMR_FRAME_88K 0x7
+#define CLASSD_INTPMR_FRAME_MASK (0x7 << 20)
+#define CLASSD_INTPMR_FRAME_SHIFT (20)
+
+#define CLASSD_INTPMR_EQCFG_FLAT 0x0
+#define CLASSD_INTPMR_EQCFG_B_BOOST_12 0x1
+#define CLASSD_INTPMR_EQCFG_B_BOOST_6 0x2
+#define CLASSD_INTPMR_EQCFG_B_CUT_12 0x3
+#define CLASSD_INTPMR_EQCFG_B_CUT_6 0x4
+#define CLASSD_INTPMR_EQCFG_M_BOOST_3 0x5
+#define CLASSD_INTPMR_EQCFG_M_BOOST_8 0x6
+#define CLASSD_INTPMR_EQCFG_M_CUT_3 0x7
+#define CLASSD_INTPMR_EQCFG_M_CUT_8 0x8
+#define CLASSD_INTPMR_EQCFG_T_BOOST_12 0x9
+#define CLASSD_INTPMR_EQCFG_T_BOOST_6 0xa
+#define CLASSD_INTPMR_EQCFG_T_CUT_12 0xb
+#define CLASSD_INTPMR_EQCFG_T_CUT_6 0xc
+#define CLASSD_INTPMR_EQCFG_SHIFT (24)
+
+#define CLASSD_INTPMR_MONO_DIS 0x0
+#define CLASSD_INTPMR_MONO_EN 0x1
+#define CLASSD_INTPMR_MONO_MASK (0x1 << 28)
+#define CLASSD_INTPMR_MONO_SHIFT (28)
+
+#define CLASSD_INTPMR_MONO_MODE_MIX 0x0
+#define CLASSD_INTPMR_MONO_MODE_SAT 0x1
+#define CLASSD_INTPMR_MONO_MODE_LEFT 0x2
+#define CLASSD_INTPMR_MONO_MODE_RIGHT 0x3
+#define CLASSD_INTPMR_MONO_MODE_MASK (0x3 << 29)
+#define CLASSD_INTPMR_MONO_MODE_SHIFT (29)
+
+#define CLASSD_INTSR 0x0000000c
+
+#define CLASSD_THR 0x00000010
+
+#define CLASSD_IER 0x00000014
+
+#define CLASSD_IDR 0x00000018
+
+#define CLASSD_IMR 0x0000001c
+
+#define CLASSD_ISR 0x00000020
+
+#define CLASSD_WPMR 0x000000e4
+
+#endif
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index aa354e1..1933bcd 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -176,6 +176,7 @@
{ .compatible = "atmel,asoc-wm8904", },
{ }
};
+MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids);
#endif
static struct platform_driver atmel_asoc_wm8904_driver = {
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
index 452f404..e97c327 100644
--- a/sound/soc/au1x/db1000.c
+++ b/sound/soc/au1x/db1000.c
@@ -38,14 +38,7 @@
{
struct snd_soc_card *card = &db1000_ac97;
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1000_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1000_audio_driver = {
@@ -54,7 +47,6 @@
.pm = &snd_soc_pm_ops,
},
.probe = db1000_audio_probe,
- .remove = db1000_audio_remove,
};
module_platform_driver(db1000_audio_driver);
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 8c907eb..5c73061 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -178,14 +178,7 @@
card = db1200_cards[pid->driver_data];
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1200_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1200_audio_driver = {
@@ -195,7 +188,6 @@
},
.id_table = db1200_pids,
.probe = db1200_audio_probe,
- .remove = db1200_audio_remove,
};
module_platform_driver(db1200_audio_driver);
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index 5bf1501..864df26 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -87,27 +87,18 @@
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "Failed to register card\n");
return ret;
}
-static int bf5xx_ad1836_driver_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver bf5xx_ad1836_driver = {
.driver = {
.name = "bfin-snd-ad1836",
.pm = &snd_soc_pm_ops,
},
.probe = bf5xx_ad1836_driver_probe,
- .remove = bf5xx_ad1836_driver_remove,
};
module_platform_driver(bf5xx_ad1836_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
index 523baf58..72ac789 100644
--- a/sound/soc/blackfin/bfin-eval-adau1373.c
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -154,16 +154,7 @@
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1373);
-}
-
-static int bfin_eval_adau1373_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373);
}
static struct platform_driver bfin_eval_adau1373_driver = {
@@ -172,7 +163,6 @@
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1373_probe,
- .remove = bfin_eval_adau1373_remove,
};
module_platform_driver(bfin_eval_adau1373_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
index f9e926d..5c67f72 100644
--- a/sound/soc/blackfin/bfin-eval-adau1701.c
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -94,16 +94,7 @@
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1701);
-}
-
-static int bfin_eval_adau1701_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701);
}
static struct platform_driver bfin_eval_adau1701_driver = {
@@ -112,7 +103,6 @@
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1701_probe,
- .remove = bfin_eval_adau1701_remove,
};
module_platform_driver(bfin_eval_adau1701_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
index 27eee66..1037477 100644
--- a/sound/soc/blackfin/bfin-eval-adav80x.c
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -119,16 +119,7 @@
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adav80x);
-}
-
-static int bfin_eval_adav80x_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x);
}
static const struct platform_device_id bfin_eval_adav80x_ids[] = {
@@ -144,7 +135,6 @@
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adav80x_probe,
- .remove = bfin_eval_adav80x_remove,
.id_table = bfin_eval_adav80x_ids,
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0c9733e..cfdafc4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
select SND_SOC_AK4554
+ select SND_SOC_AK4613 if I2C
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
@@ -57,6 +58,7 @@
select SND_SOC_CX20442 if TTY
select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
select SND_SOC_DA7213 if I2C
+ select SND_SOC_DA7219 if I2C
select SND_SOC_DA732X if I2C
select SND_SOC_DA9055 if I2C
select SND_SOC_DMIC
@@ -79,7 +81,7 @@
select SND_SOC_MAX9877 if I2C
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
- select SND_SOC_HDMI_CODEC
+ select SND_SOC_NAU8825 if I2C
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
@@ -171,6 +173,7 @@
select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8996 if I2C
select SND_SOC_WM8997 if MFD_WM8997
+ select SND_SOC_WM8998 if MFD_WM8998
select SND_SOC_WM9081 if I2C
select SND_SOC_WM9090 if I2C
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
@@ -195,9 +198,11 @@
default y if SND_SOC_WM5102=y
default y if SND_SOC_WM5110=y
default y if SND_SOC_WM8997=y
+ default y if SND_SOC_WM8998=y
default m if SND_SOC_WM5102=m
default m if SND_SOC_WM5110=m
default m if SND_SOC_WM8997=m
+ default m if SND_SOC_WM8998=m
config SND_SOC_WM_HUBS
tristate
@@ -319,6 +324,10 @@
config SND_SOC_AK4554
tristate "AKM AK4554 CODEC"
+config SND_SOC_AK4613
+ tristate "AKM AK4613 CODEC"
+ depends on I2C
+
config SND_SOC_AK4641
tristate
@@ -430,6 +439,9 @@
config SND_SOC_DA7213
tristate
+config SND_SOC_DA7219
+ tristate
+
config SND_SOC_DA732X
tristate
@@ -442,9 +454,6 @@
config SND_SOC_DMIC
tristate
-config SND_SOC_HDMI_CODEC
- tristate "HDMI stub CODEC"
-
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
@@ -865,6 +874,9 @@
config SND_SOC_WM8997
tristate
+config SND_SOC_WM8998
+ tristate
+
config SND_SOC_WM9081
tristate
@@ -896,6 +908,9 @@
config SND_SOC_ML26124
tristate
+config SND_SOC_NAU8825
+ tristate
+
config SND_SOC_TPA6130A2
tristate "Texas Instruments TPA6130A2 headphone amplifier"
depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4a32077..f632fc4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -26,6 +26,7 @@
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4554-objs := ak4554.o
+snd-soc-ak4613-objs := ak4613.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
@@ -49,6 +50,7 @@
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
snd-soc-da7213-objs := da7213.o
+snd-soc-da7219-objs := da7219.o da7219-aad.o
snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o
snd-soc-bt-sco-objs := bt-sco.o
@@ -72,7 +74,7 @@
snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
-snd-soc-hdmi-codec-objs := hdmi.o
+snd-soc-nau8825-objs := nau8825.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
@@ -176,6 +178,7 @@
snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o
snd-soc-wm8995-objs := wm8995.o
snd-soc-wm8997-objs := wm8997.o
+snd-soc-wm8998-objs := wm8998.o
snd-soc-wm9081-objs := wm9081.o
snd-soc-wm9090-objs := wm9090.o
snd-soc-wm9705-objs := wm9705.o
@@ -216,6 +219,7 @@
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o
+obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o
obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
@@ -241,6 +245,7 @@
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o
+obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o
obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
@@ -264,7 +269,7 @@
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
-obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
+obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
@@ -364,6 +369,7 @@
obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o
obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o
obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o
+obj-$(CONFIG_SND_SOC_WM8998) += snd-soc-wm8998.o
obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o
obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c
index df3a1a4..1713136 100644
--- a/sound/soc/codecs/ad193x-i2c.c
+++ b/sound/soc/codecs/ad193x-i2c.c
@@ -15,8 +15,8 @@
#include "ad193x.h"
static const struct i2c_device_id ad193x_id[] = {
- { "ad1936", 0 },
- { "ad1937", 0 },
+ { "ad1936", AD193X },
+ { "ad1937", AD193X },
{ }
};
MODULE_DEVICE_TABLE(i2c, ad193x_id);
@@ -30,7 +30,9 @@
config.val_bits = 8;
config.reg_bits = 8;
- return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config));
+ return ad193x_probe(&client->dev,
+ devm_regmap_init_i2c(client, &config),
+ (enum ad193x_type)id->driver_data);
}
static int ad193x_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c
index 8199a3d..23c2857 100644
--- a/sound/soc/codecs/ad193x-spi.c
+++ b/sound/soc/codecs/ad193x-spi.c
@@ -16,6 +16,7 @@
static int ad193x_spi_probe(struct spi_device *spi)
{
+ const struct spi_device_id *id = spi_get_device_id(spi);
struct regmap_config config;
config = ad193x_regmap_config;
@@ -24,7 +25,8 @@
config.read_flag_mask = 0x09;
config.write_flag_mask = 0x08;
- return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+ return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config),
+ (enum ad193x_type)id->driver_data);
}
static int ad193x_spi_remove(struct spi_device *spi)
@@ -33,12 +35,24 @@
return 0;
}
+static const struct spi_device_id ad193x_spi_id[] = {
+ { "ad193x", AD193X },
+ { "ad1933", AD1933 },
+ { "ad1934", AD1934 },
+ { "ad1938", AD193X },
+ { "ad1939", AD193X },
+ { "adau1328", AD193X },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, ad193x_spi_id);
+
static struct spi_driver ad193x_spi_driver = {
.driver = {
.name = "ad193x",
},
.probe = ad193x_spi_probe,
.remove = ad193x_spi_remove,
+ .id_table = ad193x_spi_id,
};
module_spi_driver(ad193x_spi_driver);
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 17c9535..3a3f3f2 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -23,6 +23,7 @@
/* codec private data */
struct ad193x_priv {
struct regmap *regmap;
+ enum ad193x_type type;
int sysclk;
};
@@ -47,12 +48,6 @@
SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL,
AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv),
- /* ADC switch control */
- SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE,
- AD193X_ADCR1_MUTE, 1, 1),
- SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE,
- AD193X_ADCR2_MUTE, 1, 1),
-
/* DAC switch control */
SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE,
AD193X_DACR1_MUTE, 1, 1),
@@ -63,26 +58,37 @@
SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE,
AD193X_DACR4_MUTE, 1, 1),
+ /* DAC de-emphasis */
+ SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum),
+};
+
+static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = {
+ /* ADC switch control */
+ SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE,
+ AD193X_ADCR1_MUTE, 1, 1),
+ SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE,
+ AD193X_ADCR2_MUTE, 1, 1),
+
/* ADC high-pass filter */
SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0,
AD193X_ADC_HIGHPASS_FILTER, 1, 0),
-
- /* DAC de-emphasis */
- SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum),
};
static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0),
- SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0),
- SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0),
SND_SOC_DAPM_VMID("VMID"),
SND_SOC_DAPM_OUTPUT("DAC1OUT"),
SND_SOC_DAPM_OUTPUT("DAC2OUT"),
SND_SOC_DAPM_OUTPUT("DAC3OUT"),
SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+};
+
+static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = {
+ SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
SND_SOC_DAPM_INPUT("ADC1IN"),
SND_SOC_DAPM_INPUT("ADC2IN"),
};
@@ -91,18 +97,33 @@
{ "DAC", NULL, "SYSCLK" },
{ "DAC Output", NULL, "DAC" },
{ "DAC Output", NULL, "VMID" },
- { "ADC", NULL, "SYSCLK" },
- { "DAC", NULL, "ADC_PWR" },
- { "ADC", NULL, "ADC_PWR" },
{ "DAC1OUT", NULL, "DAC Output" },
{ "DAC2OUT", NULL, "DAC Output" },
{ "DAC3OUT", NULL, "DAC Output" },
{ "DAC4OUT", NULL, "DAC Output" },
- { "ADC", NULL, "ADC1IN" },
- { "ADC", NULL, "ADC2IN" },
{ "SYSCLK", NULL, "PLL_PWR" },
};
+static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = {
+ { "ADC", NULL, "SYSCLK" },
+ { "ADC", NULL, "ADC_PWR" },
+ { "ADC", NULL, "ADC1IN" },
+ { "ADC", NULL, "ADC2IN" },
+};
+
+static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x)
+{
+ switch (ad193x->type) {
+ case AD1933:
+ case AD1934:
+ return false;
+ default:
+ break;
+ }
+
+ return true;
+}
+
/*
* DAI ops entries
*/
@@ -147,8 +168,10 @@
regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1,
AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT);
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
- AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT);
+ if (ad193x_has_adc(ad193x))
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
+ AD193X_ADC_CHAN_MASK,
+ channels << AD193X_ADC_CHAN_SHFT);
return 0;
}
@@ -172,7 +195,9 @@
adc_serfmt |= AD193X_ADC_SERFMT_AUX;
break;
default:
- return -EINVAL;
+ if (ad193x_has_adc(ad193x))
+ return -EINVAL;
+ break;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -217,10 +242,12 @@
return -EINVAL;
}
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
- AD193X_ADC_SERFMT_MASK, adc_serfmt);
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
- AD193X_ADC_FMT_MASK, adc_fmt);
+ if (ad193x_has_adc(ad193x)) {
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
+ AD193X_ADC_SERFMT_MASK, adc_serfmt);
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
+ AD193X_ADC_FMT_MASK, adc_fmt);
+ }
regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1,
AD193X_DAC_FMT_MASK, dac_fmt);
@@ -287,8 +314,9 @@
AD193X_DAC_WORD_LEN_MASK,
word_len << AD193X_DAC_WORD_LEN_SHFT);
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
- AD193X_ADC_WORD_LEN_MASK, word_len);
+ if (ad193x_has_adc(ad193x))
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
+ AD193X_ADC_WORD_LEN_MASK, word_len);
return 0;
}
@@ -326,6 +354,8 @@
static int ad193x_codec_probe(struct snd_soc_codec *codec)
{
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ int num, ret;
/* default setting for ad193x */
@@ -335,14 +365,46 @@
regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A);
/* dac in tdm mode */
regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40);
- /* high-pass filter enable */
- regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
- /* sata delay=1, adc aux mode */
- regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
+
+ /* adc only */
+ if (ad193x_has_adc(ad193x)) {
+ /* high-pass filter enable */
+ regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
+ /* sata delay=1, adc aux mode */
+ regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
+ }
+
/* pll input: mclki/xi */
regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04);
+ /* adc only */
+ if (ad193x_has_adc(ad193x)) {
+ /* add adc controls */
+ num = ARRAY_SIZE(ad193x_adc_snd_controls);
+ ret = snd_soc_add_codec_controls(codec,
+ ad193x_adc_snd_controls,
+ num);
+ if (ret)
+ return ret;
+
+ /* add adc widgets */
+ num = ARRAY_SIZE(ad193x_adc_widgets);
+ ret = snd_soc_dapm_new_controls(dapm,
+ ad193x_adc_widgets,
+ num);
+ if (ret)
+ return ret;
+
+ /* add adc routes */
+ num = ARRAY_SIZE(ad193x_adc_audio_paths);
+ ret = snd_soc_dapm_add_routes(dapm,
+ ad193x_adc_audio_paths,
+ num);
+ if (ret)
+ return ret;
+ }
+
return 0;
}
@@ -356,18 +418,13 @@
.num_dapm_routes = ARRAY_SIZE(audio_paths),
};
-static bool adau193x_reg_volatile(struct device *dev, unsigned int reg)
-{
- return false;
-}
-
const struct regmap_config ad193x_regmap_config = {
.max_register = AD193X_NUM_REGS - 1,
- .volatile_reg = adau193x_reg_volatile,
};
EXPORT_SYMBOL_GPL(ad193x_regmap_config);
-int ad193x_probe(struct device *dev, struct regmap *regmap)
+int ad193x_probe(struct device *dev, struct regmap *regmap,
+ enum ad193x_type type)
{
struct ad193x_priv *ad193x;
@@ -379,6 +436,7 @@
return -ENOMEM;
ad193x->regmap = regmap;
+ ad193x->type = type;
dev_set_drvdata(dev, ad193x);
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index ab9a998..8b1e65f 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -13,8 +13,15 @@
struct device;
+enum ad193x_type {
+ AD193X,
+ AD1933,
+ AD1934,
+};
+
extern const struct regmap_config ad193x_regmap_config;
-int ad193x_probe(struct device *dev, struct regmap *regmap);
+int ad193x_probe(struct device *dev, struct regmap *regmap,
+ enum ad193x_type type);
#define AD193X_PLL_CLK_CTRL0 0x00
#define AD193X_PLL_POWERDOWN 0x01
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 198c924..acff8d6 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -728,8 +728,8 @@
if (!snd_soc_codec_is_active(codec) || !adav80x->rate)
return 0;
- return snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+ return snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, adav80x->rate);
}
static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
new file mode 100644
index 0000000..07a2664
--- /dev/null
+++ b/sound/soc/codecs/ak4613.c
@@ -0,0 +1,497 @@
+/*
+ * ak4613.c -- Asahi Kasei ALSA Soc Audio driver
+ *
+ * Copyright (C) 2015 Renesas Electronics Corporation
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on ak4642.c by Kuninori Morimoto
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * 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.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#define PW_MGMT1 0x00 /* Power Management 1 */
+#define PW_MGMT2 0x01 /* Power Management 2 */
+#define PW_MGMT3 0x02 /* Power Management 3 */
+#define CTRL1 0x03 /* Control 1 */
+#define CTRL2 0x04 /* Control 2 */
+#define DEMP1 0x05 /* De-emphasis1 */
+#define DEMP2 0x06 /* De-emphasis2 */
+#define OFD 0x07 /* Overflow Detect */
+#define ZRD 0x08 /* Zero Detect */
+#define ICTRL 0x09 /* Input Control */
+#define OCTRL 0x0a /* Output Control */
+#define LOUT1 0x0b /* LOUT1 Volume Control */
+#define ROUT1 0x0c /* ROUT1 Volume Control */
+#define LOUT2 0x0d /* LOUT2 Volume Control */
+#define ROUT2 0x0e /* ROUT2 Volume Control */
+#define LOUT3 0x0f /* LOUT3 Volume Control */
+#define ROUT3 0x10 /* ROUT3 Volume Control */
+#define LOUT4 0x11 /* LOUT4 Volume Control */
+#define ROUT4 0x12 /* ROUT4 Volume Control */
+#define LOUT5 0x13 /* LOUT5 Volume Control */
+#define ROUT5 0x14 /* ROUT5 Volume Control */
+#define LOUT6 0x15 /* LOUT6 Volume Control */
+#define ROUT6 0x16 /* ROUT6 Volume Control */
+
+/* PW_MGMT1 */
+#define RSTN BIT(0)
+#define PMDAC BIT(1)
+#define PMADC BIT(2)
+#define PMVR BIT(3)
+
+/* PW_MGMT2 */
+#define PMAD_ALL 0x7
+
+/* PW_MGMT3 */
+#define PMDA_ALL 0x3f
+
+/* CTRL1 */
+#define DIF0 BIT(3)
+#define DIF1 BIT(4)
+#define DIF2 BIT(5)
+#define TDM0 BIT(6)
+#define TDM1 BIT(7)
+#define NO_FMT (0xff)
+#define FMT_MASK (0xf8)
+
+/* CTRL2 */
+#define DFS_NORMAL_SPEED (0 << 2)
+#define DFS_DOUBLE_SPEED (1 << 2)
+#define DFS_QUAD_SPEED (2 << 2)
+
+struct ak4613_priv {
+ struct mutex lock;
+
+ unsigned int fmt;
+ u8 fmt_ctrl;
+ int cnt;
+};
+
+struct ak4613_formats {
+ unsigned int width;
+ unsigned int fmt;
+};
+
+struct ak4613_interface {
+ struct ak4613_formats capture;
+ struct ak4613_formats playback;
+};
+
+/*
+ * Playback Volume
+ *
+ * max : 0x00 : 0 dB
+ * ( 0.5 dB step )
+ * min : 0xFE : -127.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new ak4613_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6,
+ 0, 0xFF, 1, out_tlv),
+};
+
+static const struct reg_default ak4613_reg[] = {
+ { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 },
+ { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 },
+ { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 },
+ { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 },
+ { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 },
+ { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
+};
+
+#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3)
+#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt }
+static const struct ak4613_interface ak4613_iface[] = {
+ /* capture */ /* playback */
+ [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) },
+ [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) },
+ [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) },
+ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) },
+ [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) },
+};
+
+static const struct regmap_config ak4613_regmap_cfg = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = 0x16,
+ .reg_defaults = ak4613_reg,
+ .num_reg_defaults = ARRAY_SIZE(ak4613_reg),
+};
+
+static const struct of_device_id ak4613_of_match[] = {
+ { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg },
+ {},
+};
+MODULE_DEVICE_TABLE(of, ak4613_of_match);
+
+static const struct i2c_device_id ak4613_i2c_id[] = {
+ { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id);
+
+static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = {
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("LOUT3"),
+ SND_SOC_DAPM_OUTPUT("LOUT4"),
+ SND_SOC_DAPM_OUTPUT("LOUT5"),
+ SND_SOC_DAPM_OUTPUT("LOUT6"),
+
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT3"),
+ SND_SOC_DAPM_OUTPUT("ROUT4"),
+ SND_SOC_DAPM_OUTPUT("ROUT5"),
+ SND_SOC_DAPM_OUTPUT("ROUT6"),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("LIN1"),
+ SND_SOC_DAPM_INPUT("LIN2"),
+
+ SND_SOC_DAPM_INPUT("RIN1"),
+ SND_SOC_DAPM_INPUT("RIN2"),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0),
+ SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0),
+ SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0),
+ SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0),
+ SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0),
+ SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0),
+ SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0),
+};
+
+static const struct snd_soc_dapm_route ak4613_intercon[] = {
+ {"LOUT1", NULL, "DAC1"},
+ {"LOUT2", NULL, "DAC2"},
+ {"LOUT3", NULL, "DAC3"},
+ {"LOUT4", NULL, "DAC4"},
+ {"LOUT5", NULL, "DAC5"},
+ {"LOUT6", NULL, "DAC6"},
+
+ {"ROUT1", NULL, "DAC1"},
+ {"ROUT2", NULL, "DAC2"},
+ {"ROUT3", NULL, "DAC3"},
+ {"ROUT4", NULL, "DAC4"},
+ {"ROUT5", NULL, "DAC5"},
+ {"ROUT6", NULL, "DAC6"},
+
+ {"DAC1", NULL, "Playback"},
+ {"DAC2", NULL, "Playback"},
+ {"DAC3", NULL, "Playback"},
+ {"DAC4", NULL, "Playback"},
+ {"DAC5", NULL, "Playback"},
+ {"DAC6", NULL, "Playback"},
+
+ {"Capture", NULL, "ADC1"},
+ {"Capture", NULL, "ADC2"},
+
+ {"ADC1", NULL, "LIN1"},
+ {"ADC2", NULL, "LIN2"},
+
+ {"ADC1", NULL, "RIN1"},
+ {"ADC2", NULL, "RIN2"},
+};
+
+static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+
+ mutex_lock(&priv->lock);
+ priv->cnt--;
+ if (priv->cnt < 0) {
+ dev_err(dev, "unexpected counter error\n");
+ priv->cnt = 0;
+ }
+ if (!priv->cnt)
+ priv->fmt_ctrl = NO_FMT;
+ mutex_unlock(&priv->lock);
+}
+
+static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ fmt &= SND_SOC_DAIFMT_FORMAT_MASK;
+
+ switch (fmt) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_I2S:
+ priv->fmt = fmt;
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ const struct ak4613_formats *fmts;
+ struct device *dev = codec->dev;
+ unsigned int width = params_width(params);
+ unsigned int fmt = priv->fmt;
+ unsigned int rate;
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ int i, ret;
+ u8 fmt_ctrl, ctrl2;
+
+ rate = params_rate(params);
+ switch (rate) {
+ case 32000:
+ case 44100:
+ case 48000:
+ ctrl2 = DFS_NORMAL_SPEED;
+ break;
+ case 88200:
+ case 96000:
+ ctrl2 = DFS_DOUBLE_SPEED;
+ break;
+ case 176400:
+ case 192000:
+ ctrl2 = DFS_QUAD_SPEED;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /*
+ * FIXME
+ *
+ * It doesn't support TDM at this point
+ */
+ fmt_ctrl = NO_FMT;
+ for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) {
+ fmts = (is_play) ? &ak4613_iface[i].playback :
+ &ak4613_iface[i].capture;
+
+ if (fmts->fmt != fmt)
+ continue;
+
+ if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
+ if (fmts->width != width)
+ continue;
+ } else {
+ if (fmts->width < width)
+ continue;
+ }
+
+ fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i);
+ break;
+ }
+
+ ret = -EINVAL;
+ if (fmt_ctrl == NO_FMT)
+ goto hw_params_end;
+
+ mutex_lock(&priv->lock);
+ if ((priv->fmt_ctrl == NO_FMT) ||
+ (priv->fmt_ctrl == fmt_ctrl)) {
+ priv->fmt_ctrl = fmt_ctrl;
+ priv->cnt++;
+ ret = 0;
+ }
+ mutex_unlock(&priv->lock);
+
+ if (ret < 0)
+ goto hw_params_end;
+
+ snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl);
+ snd_soc_write(codec, CTRL2, ctrl2);
+
+hw_params_end:
+ if (ret < 0)
+ dev_warn(dev, "unsupported data width/format combination\n");
+
+ return ret;
+}
+
+static int ak4613_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 mgmt1 = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ mgmt1 |= RSTN;
+ /* fall through */
+ case SND_SOC_BIAS_PREPARE:
+ mgmt1 |= PMADC | PMDAC;
+ /* fall through */
+ case SND_SOC_BIAS_STANDBY:
+ mgmt1 |= PMVR;
+ /* fall through */
+ case SND_SOC_BIAS_OFF:
+ default:
+ break;
+ }
+
+ snd_soc_write(codec, PW_MGMT1, mgmt1);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops ak4613_dai_ops = {
+ .shutdown = ak4613_dai_shutdown,
+ .set_fmt = ak4613_dai_set_fmt,
+ .hw_params = ak4613_dai_hw_params,
+};
+
+#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_64000 |\
+ SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000 |\
+ SNDRV_PCM_RATE_176400 |\
+ SNDRV_PCM_RATE_192000)
+#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver ak4613_dai = {
+ .name = "ak4613-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .ops = &ak4613_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+ struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+ regcache_mark_dirty(regmap);
+ return regcache_sync(regmap);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+ .resume = ak4613_resume,
+ .set_bias_level = ak4613_set_bias_level,
+ .controls = ak4613_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4613_snd_controls),
+ .dapm_widgets = ak4613_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets),
+ .dapm_routes = ak4613_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ak4613_intercon),
+};
+
+static int ak4613_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
+ const struct regmap_config *regmap_cfg;
+ struct regmap *regmap;
+ struct ak4613_priv *priv;
+
+ regmap_cfg = NULL;
+ if (np) {
+ const struct of_device_id *of_id;
+
+ of_id = of_match_device(ak4613_of_match, dev);
+ if (of_id)
+ regmap_cfg = of_id->data;
+ } else {
+ regmap_cfg = (const struct regmap_config *)id->driver_data;
+ }
+
+ if (!regmap_cfg)
+ return -EINVAL;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->fmt_ctrl = NO_FMT;
+ priv->cnt = 0;
+
+ mutex_init(&priv->lock);
+
+ i2c_set_clientdata(i2c, priv);
+
+ regmap = devm_regmap_init_i2c(i2c, regmap_cfg);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ return snd_soc_register_codec(dev, &soc_codec_dev_ak4613,
+ &ak4613_dai, 1);
+}
+
+static int ak4613_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver ak4613_i2c_driver = {
+ .driver = {
+ .name = "ak4613-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = ak4613_of_match,
+ },
+ .probe = ak4613_i2c_probe,
+ .remove = ak4613_i2c_remove,
+ .id_table = ak4613_i2c_id,
+};
+
+module_i2c_driver(ak4613_i2c_driver);
+
+MODULE_DESCRIPTION("Soc AK4613 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4a90143..cda27c2 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -23,6 +23,8 @@
* AK4648 is tested.
*/
+#include <linux/clk.h>
+#include <linux/clk-provider.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
@@ -128,11 +130,8 @@
#define I2S (3 << 0)
/* MD_CTL2 */
-#define FS0 (1 << 0)
-#define FS1 (1 << 1)
-#define FS2 (1 << 2)
-#define FS3 (1 << 5)
-#define FS_MASK (FS0 | FS1 | FS2 | FS3)
+#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2))
+#define PSs(val) ((val & 0x3) << 6)
/* MD_CTL3 */
#define BST1 (1 << 3)
@@ -147,6 +146,7 @@
struct ak4642_priv {
const struct ak4642_drvdata *drvdata;
+ struct clk *mcko;
};
/*
@@ -430,56 +430,56 @@
return 0;
}
+static int ak4642_set_mcko(struct snd_soc_codec *codec,
+ u32 frequency)
+{
+ u32 fs_list[] = {
+ [0] = 8000,
+ [1] = 12000,
+ [2] = 16000,
+ [3] = 24000,
+ [4] = 7350,
+ [5] = 11025,
+ [6] = 14700,
+ [7] = 22050,
+ [10] = 32000,
+ [11] = 48000,
+ [14] = 29400,
+ [15] = 44100,
+ };
+ u32 ps_list[] = {
+ [0] = 256,
+ [1] = 128,
+ [2] = 64,
+ [3] = 32
+ };
+ int ps, fs;
+
+ for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
+ for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
+ if (frequency == ps_list[ps] * fs_list[fs]) {
+ snd_soc_write(codec, MD_CTL2,
+ PSs(ps) | FSs(fs));
+ return 0;
+ }
+ }
+ }
+
+ return 0;
+}
+
static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
- u8 rate;
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+ u32 rate = clk_get_rate(priv->mcko);
- switch (params_rate(params)) {
- case 7350:
- rate = FS2;
- break;
- case 8000:
- rate = 0;
- break;
- case 11025:
- rate = FS2 | FS0;
- break;
- case 12000:
- rate = FS0;
- break;
- case 14700:
- rate = FS2 | FS1;
- break;
- case 16000:
- rate = FS1;
- break;
- case 22050:
- rate = FS2 | FS1 | FS0;
- break;
- case 24000:
- rate = FS1 | FS0;
- break;
- case 29400:
- rate = FS3 | FS2 | FS1;
- break;
- case 32000:
- rate = FS3 | FS1;
- break;
- case 44100:
- rate = FS3 | FS2 | FS1 | FS0;
- break;
- case 48000:
- rate = FS3 | FS1 | FS0;
- break;
- default:
- return -EINVAL;
- }
- snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
+ if (!rate)
+ rate = params_rate(params) * 256;
- return 0;
+ return ak4642_set_mcko(codec, rate);
}
static int ak4642_set_bias_level(struct snd_soc_codec *codec,
@@ -532,7 +532,18 @@
return 0;
}
+static int ak4642_probe(struct snd_soc_codec *codec)
+{
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (priv->mcko)
+ ak4642_set_mcko(codec, clk_get_rate(priv->mcko));
+
+ return 0;
+}
+
static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+ .probe = ak4642_probe,
.resume = ak4642_resume,
.set_bias_level = ak4642_set_bias_level,
.controls = ak4642_snd_controls,
@@ -580,19 +591,54 @@
.extended_frequencies = 1,
};
+#ifdef CONFIG_COMMON_CLK
+static struct clk *ak4642_of_parse_mcko(struct device *dev)
+{
+ struct device_node *np = dev->of_node;
+ struct clk *clk;
+ const char *clk_name = np->name;
+ const char *parent_clk_name = NULL;
+ u32 rate;
+
+ if (of_property_read_u32(np, "clock-frequency", &rate))
+ return NULL;
+
+ if (of_property_read_bool(np, "clocks"))
+ parent_clk_name = of_clk_get_parent_name(np, 0);
+
+ of_property_read_string(np, "clock-output-names", &clk_name);
+
+ clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
+ (parent_clk_name) ? 0 : CLK_IS_ROOT,
+ rate);
+ if (!IS_ERR(clk))
+ of_clk_add_provider(np, of_clk_src_simple_get, clk);
+
+ return clk;
+}
+#else
+#define ak4642_of_parse_mcko(d) 0
+#endif
+
static const struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
- struct device_node *np = i2c->dev.of_node;
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
const struct ak4642_drvdata *drvdata = NULL;
struct regmap *regmap;
struct ak4642_priv *priv;
+ struct clk *mcko = NULL;
if (np) {
const struct of_device_id *of_id;
- of_id = of_match_device(ak4642_of_match, &i2c->dev);
+ mcko = ak4642_of_parse_mcko(dev);
+ if (IS_ERR(mcko))
+ mcko = NULL;
+
+ of_id = of_match_device(ak4642_of_match, dev);
if (of_id)
drvdata = of_id->data;
} else {
@@ -600,15 +646,16 @@
}
if (!drvdata) {
- dev_err(&i2c->dev, "Unknown device type\n");
+ dev_err(dev, "Unknown device type\n");
return -EINVAL;
}
- priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
priv->drvdata = drvdata;
+ priv->mcko = mcko;
i2c_set_clientdata(i2c, priv);
@@ -616,7 +663,7 @@
if (IS_ERR(regmap))
return PTR_ERR(regmap);
- return snd_soc_register_codec(&i2c->dev,
+ return snd_soc_register_codec(dev,
&soc_codec_dev_ak4642, &ak4642_dai, 1);
}
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8a2221a..9929efc 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -147,6 +147,8 @@
0x4f5, 0x0da);
}
break;
+ default:
+ break;
}
return 0;
@@ -314,6 +316,7 @@
"Tone Generator 2",
"Haptics",
"AEC",
+ "AEC2",
"Mic Mute Mixer",
"Noise Generator",
"IN1L",
@@ -421,6 +424,7 @@
0x05,
0x06, /* Haptics */
0x08, /* AEC */
+ 0x09, /* AEC2 */
0x0c, /* Noise mixer */
0x0d, /* Comfort noise */
0x10, /* IN1L */
@@ -525,6 +529,32 @@
const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
+const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = {
+ "12kHz", "24kHz", "48kHz", "96kHz", "192kHz",
+ "11.025kHz", "22.05kHz", "44.1kHz", "88.2kHz", "176.4kHz",
+ "4kHz", "8kHz", "16kHz", "32kHz",
+};
+EXPORT_SYMBOL_GPL(arizona_sample_rate_text);
+
+const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = {
+ 0x01, 0x02, 0x03, 0x04, 0x05, 0x09, 0x0A, 0x0B, 0x0C, 0x0D,
+ 0x10, 0x11, 0x12, 0x13,
+};
+EXPORT_SYMBOL_GPL(arizona_sample_rate_val);
+
+const char *arizona_sample_rate_val_to_name(unsigned int rate_val)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(arizona_sample_rate_val); ++i) {
+ if (arizona_sample_rate_val[i] == rate_val)
+ return arizona_sample_rate_text[i];
+ }
+
+ return "Illegal";
+}
+EXPORT_SYMBOL_GPL(arizona_sample_rate_val_to_name);
+
const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = {
"SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate",
};
@@ -689,6 +719,15 @@
ARIZONA_IN_VU, val);
}
+bool arizona_input_analog(struct snd_soc_codec *codec, int shift)
+{
+ unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8);
+ unsigned int val = snd_soc_read(codec, reg);
+
+ return !(val & ARIZONA_IN1_MODE_MASK);
+}
+EXPORT_SYMBOL_GPL(arizona_input_analog);
+
int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
int event)
{
@@ -725,6 +764,9 @@
reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES);
if (reg == 0)
arizona_in_set_vu(codec, 0);
+ break;
+ default:
+ break;
}
return 0;
@@ -806,6 +848,8 @@
break;
}
break;
+ default:
+ break;
}
return 0;
@@ -1868,6 +1912,11 @@
if (fll->arizona->rev < 3 || sync)
return init_ratio;
break;
+ case WM8998:
+ case WM1814:
+ if (sync)
+ return init_ratio;
+ break;
default:
return init_ratio;
}
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ada0a41..fea8b8a 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -93,12 +93,17 @@
bool dvfs_cached;
};
-#define ARIZONA_NUM_MIXER_INPUTS 103
+#define ARIZONA_NUM_MIXER_INPUTS 104
extern const unsigned int arizona_mixer_tlv[];
extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
+#define ARIZONA_GAINMUX_CONTROLS(name, base) \
+ SOC_SINGLE_RANGE_TLV(name " Input Volume", base + 1, \
+ ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
+ arizona_mixer_tlv)
+
#define ARIZONA_MIXER_CONTROLS(name, base) \
SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \
ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
@@ -209,8 +214,12 @@
.num_regs = 1 }) }
#define ARIZONA_RATE_ENUM_SIZE 4
+#define ARIZONA_SAMPLE_RATE_ENUM_SIZE 14
+
extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
+extern const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
+extern const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
extern const struct soc_enum arizona_isrc_fsl[];
extern const struct soc_enum arizona_isrc_fsh[];
@@ -294,4 +303,7 @@
int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
bool diff);
+extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
+
+extern const char *arizona_sample_rate_val_to_name(unsigned int rate_val);
#endif
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index a9c86ef..7278f93 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -12,6 +12,7 @@
* option) any later version.
*/
+#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
@@ -1222,23 +1223,44 @@
{
struct snd_soc_codec *codec = codec_dai->codec;
struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ if ((da7213->clk_src == clk_id) && (da7213->mclk_rate == freq))
+ return 0;
+
+ if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) {
+ dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+ freq);
+ return -EINVAL;
+ }
switch (clk_id) {
case DA7213_CLKSRC_MCLK:
- if ((freq == 32768) ||
- ((freq >= 5000000) && (freq <= 54000000))) {
- da7213->mclk_rate = freq;
- return 0;
- } else {
- dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
- freq);
- return -EINVAL;
- }
+ da7213->mclk_squarer_en = false;
+ break;
+ case DA7213_CLKSRC_MCLK_SQR:
+ da7213->mclk_squarer_en = true;
break;
default:
dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
return -EINVAL;
}
+
+ da7213->clk_src = clk_id;
+
+ if (da7213->mclk) {
+ freq = clk_round_rate(da7213->mclk, freq);
+ ret = clk_set_rate(da7213->mclk, freq);
+ if (ret) {
+ dev_err(codec_dai->dev, "Failed to set clock rate %d\n",
+ freq);
+ return ret;
+ }
+ }
+
+ da7213->mclk_rate = freq;
+
+ return 0;
}
/* Supported PLL input frequencies are 5MHz - 54MHz. */
@@ -1366,12 +1388,25 @@
static int da7213_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+ /* MCLK */
+ if (da7213->mclk) {
+ ret = clk_prepare_enable(da7213->mclk);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to enable mclk\n");
+ return ret;
+ }
+ }
+
/* Enable VMID reference & master bias */
snd_soc_update_bits(codec, DA7213_REFERENCES,
DA7213_VMID_EN | DA7213_BIAS_EN,
@@ -1382,15 +1417,127 @@
/* Disable VMID reference & master bias */
snd_soc_update_bits(codec, DA7213_REFERENCES,
DA7213_VMID_EN | DA7213_BIAS_EN, 0);
+
+ /* MCLK */
+ if (da7213->mclk)
+ clk_disable_unprepare(da7213->mclk);
break;
}
return 0;
}
+/* DT */
+static const struct of_device_id da7213_of_match[] = {
+ { .compatible = "dlg,da7213", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, da7213_of_match);
+
+static enum da7213_micbias_voltage
+ da7213_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1600:
+ return DA7213_MICBIAS_1_6V;
+ case 2200:
+ return DA7213_MICBIAS_2_2V;
+ case 2500:
+ return DA7213_MICBIAS_2_5V;
+ case 3000:
+ return DA7213_MICBIAS_3_0V;
+ default:
+ dev_warn(codec->dev, "Invalid micbias level\n");
+ return DA7213_MICBIAS_2_2V;
+ }
+}
+
+static enum da7213_dmic_data_sel
+ da7213_of_dmic_data_sel(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "lrise_rfall")) {
+ return DA7213_DMIC_DATA_LRISE_RFALL;
+ } else if (!strcmp(str, "lfall_rrise")) {
+ return DA7213_DMIC_DATA_LFALL_RRISE;
+ } else {
+ dev_warn(codec->dev, "Invalid DMIC data select type\n");
+ return DA7213_DMIC_DATA_LRISE_RFALL;
+ }
+}
+
+static enum da7213_dmic_samplephase
+ da7213_of_dmic_samplephase(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "on_clkedge")) {
+ return DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+ } else if (!strcmp(str, "between_clkedge")) {
+ return DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE;
+ } else {
+ dev_warn(codec->dev, "Invalid DMIC sample phase\n");
+ return DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+ }
+}
+
+static enum da7213_dmic_clk_rate
+ da7213_of_dmic_clkrate(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1500000:
+ return DA7213_DMIC_CLK_1_5MHZ;
+ case 3000000:
+ return DA7213_DMIC_CLK_3_0MHZ;
+ default:
+ dev_warn(codec->dev, "Invalid DMIC clock rate\n");
+ return DA7213_DMIC_CLK_1_5MHZ;
+ }
+}
+
+static struct da7213_platform_data
+ *da7213_of_to_pdata(struct snd_soc_codec *codec)
+{
+ struct device_node *np = codec->dev->of_node;
+ struct da7213_platform_data *pdata;
+ const char *of_str;
+ u32 of_val32;
+
+ pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata) {
+ dev_warn(codec->dev, "Failed to allocate memory for pdata\n");
+ return NULL;
+ }
+
+ if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0)
+ pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32);
+ else
+ pdata->micbias1_lvl = DA7213_MICBIAS_2_2V;
+
+ if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0)
+ pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32);
+ else
+ pdata->micbias2_lvl = DA7213_MICBIAS_2_2V;
+
+ if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str))
+ pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str);
+ else
+ pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL;
+
+ if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str))
+ pdata->dmic_samplephase =
+ da7213_of_dmic_samplephase(codec, of_str);
+ else
+ pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+
+ if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0)
+ pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32);
+ else
+ pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ;
+
+ return pdata;
+}
+
+
static int da7213_probe(struct snd_soc_codec *codec)
{
struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
- struct da7213_platform_data *pdata = da7213->pdata;
/* Default to using ALC auto offset calibration mode. */
snd_soc_update_bits(codec, DA7213_ALC_CTRL1,
@@ -1450,8 +1597,15 @@
snd_soc_update_bits(codec, DA7213_LINE_CTRL,
DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE);
+ /* Handle DT/Platform data */
+ if (codec->dev->of_node)
+ da7213->pdata = da7213_of_to_pdata(codec);
+ else
+ da7213->pdata = dev_get_platdata(codec->dev);
+
/* Set platform data values */
if (da7213->pdata) {
+ struct da7213_platform_data *pdata = da7213->pdata;
u8 micbias_lvl = 0, dmic_cfg = 0;
/* Set Mic Bias voltages */
@@ -1503,10 +1657,17 @@
DA7213_DMIC_DATA_SEL_MASK |
DA7213_DMIC_SAMPLEPHASE_MASK |
DA7213_DMIC_CLK_RATE_MASK, dmic_cfg);
-
- /* Set MCLK squaring */
- da7213->mclk_squarer_en = pdata->mclk_squaring;
}
+
+ /* Check if MCLK provided */
+ da7213->mclk = devm_clk_get(codec->dev, "mclk");
+ if (IS_ERR(da7213->mclk)) {
+ if (PTR_ERR(da7213->mclk) != -ENOENT)
+ return PTR_ERR(da7213->mclk);
+ else
+ da7213->mclk = NULL;
+ }
+
return 0;
}
@@ -1537,7 +1698,6 @@
const struct i2c_device_id *id)
{
struct da7213_priv *da7213;
- struct da7213_platform_data *pdata = dev_get_platdata(&i2c->dev);
int ret;
da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv),
@@ -1545,9 +1705,6 @@
if (!da7213)
return -ENOMEM;
- if (pdata)
- da7213->pdata = pdata;
-
i2c_set_clientdata(i2c, da7213);
da7213->regmap = devm_regmap_init_i2c(i2c, &da7213_regmap_config);
@@ -1582,6 +1739,7 @@
static struct i2c_driver da7213_i2c_driver = {
.driver = {
.name = "da7213",
+ .of_match_table = of_match_ptr(da7213_of_match),
},
.probe = da7213_i2c_probe,
.remove = da7213_remove,
diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h
index 9cb9ddd..030fd69 100644
--- a/sound/soc/codecs/da7213.h
+++ b/sound/soc/codecs/da7213.h
@@ -13,6 +13,7 @@
#ifndef _DA7213_H
#define _DA7213_H
+#include <linux/clk.h>
#include <linux/regmap.h>
#include <sound/da7213.h>
@@ -504,14 +505,17 @@
#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8
#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16
-enum clk_src {
- DA7213_CLKSRC_MCLK
+enum da7213_clk_src {
+ DA7213_CLKSRC_MCLK = 0,
+ DA7213_CLKSRC_MCLK_SQR,
};
/* Codec private data */
struct da7213_priv {
struct regmap *regmap;
+ struct clk *mclk;
unsigned int mclk_rate;
+ int clk_src;
bool master;
bool mclk_squarer_en;
bool srm_en;
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
new file mode 100644
index 0000000..9459593
--- /dev/null
+++ b/sound/soc/codecs/da7219-aad.c
@@ -0,0 +1,823 @@
+/*
+ * da7219-aad.c - Dialog DA7219 ALSA SoC AAD Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor Ltd.
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
+#include <linux/pm_wakeirq.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/workqueue.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/da7219.h>
+
+#include "da7219.h"
+#include "da7219-aad.h"
+
+
+/*
+ * Detection control
+ */
+
+void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ da7219->aad->jack = jack;
+ da7219->aad->jack_inserted = false;
+
+ /* Send an initial empty report */
+ snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK);
+
+ /* Enable/Disable jack detection */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_ACCDET_EN_MASK,
+ (jack ? DA7219_ACCDET_EN_MASK : 0));
+}
+EXPORT_SYMBOL_GPL(da7219_aad_jack_det);
+
+/*
+ * Button/HPTest work
+ */
+
+static void da7219_aad_btn_det_work(struct work_struct *work)
+{
+ struct da7219_aad_priv *da7219_aad =
+ container_of(work, struct da7219_aad_priv, btn_det_work);
+ struct snd_soc_codec *codec = da7219_aad->codec;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ u8 statusa, micbias_ctrl;
+ bool micbias_up = false;
+ int retries = 0;
+
+ /* Drive headphones/lineout */
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_OE_MASK,
+ DA7219_HP_L_AMP_OE_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_OE_MASK,
+ DA7219_HP_R_AMP_OE_MASK);
+
+ /* Make sure mic bias is up */
+ snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+ snd_soc_dapm_sync(dapm);
+
+ do {
+ statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A);
+ if (statusa & DA7219_MICBIAS_UP_STS_MASK)
+ micbias_up = true;
+ else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES)
+ msleep(DA7219_AAD_MICBIAS_CHK_DELAY);
+ } while ((!micbias_up) && (retries < DA7219_AAD_MICBIAS_CHK_RETRIES));
+
+ if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES)
+ dev_warn(codec->dev, "Mic bias status check timed out");
+
+ /*
+ * Mic bias pulse required to enable mic, must be done before enabling
+ * button detection to prevent erroneous button readings.
+ */
+ if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) {
+ /* Pulse higher level voltage */
+ micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL);
+ snd_soc_update_bits(codec, DA7219_MICBIAS_CTRL,
+ DA7219_MICBIAS1_LEVEL_MASK,
+ da7219_aad->micbias_pulse_lvl);
+ msleep(da7219_aad->micbias_pulse_time);
+ snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_ctrl);
+
+ }
+
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_BUTTON_CONFIG_MASK,
+ da7219_aad->btn_cfg);
+}
+
+static void da7219_aad_hptest_work(struct work_struct *work)
+{
+ struct da7219_aad_priv *da7219_aad =
+ container_of(work, struct da7219_aad_priv, hptest_work);
+ struct snd_soc_codec *codec = da7219_aad->codec;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ u16 tonegen_freq_hptest;
+ u8 accdet_cfg8;
+ int report = 0;
+
+ /* Lock DAPM and any Kcontrols that are affected by this test */
+ snd_soc_dapm_mutex_lock(dapm);
+ mutex_lock(&da7219->lock);
+
+ /* Bypass cache so it saves current settings */
+ regcache_cache_bypass(da7219->regmap, true);
+
+ /* Make sure Tone Generator is disabled */
+ snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0);
+
+ /* Enable HPTest block, 1KOhms check */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8,
+ DA7219_HPTEST_EN_MASK | DA7219_HPTEST_RES_SEL_MASK,
+ DA7219_HPTEST_EN_MASK |
+ DA7219_HPTEST_RES_SEL_1KOHMS);
+
+ /* Set gains to 0db */
+ snd_soc_write(codec, DA7219_DAC_L_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB);
+ snd_soc_write(codec, DA7219_DAC_R_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB);
+ snd_soc_write(codec, DA7219_HP_L_GAIN, DA7219_HP_AMP_GAIN_0DB);
+ snd_soc_write(codec, DA7219_HP_R_GAIN, DA7219_HP_AMP_GAIN_0DB);
+
+ /* Disable DAC filters, EQs and soft mute */
+ snd_soc_update_bits(codec, DA7219_DAC_FILTERS1, DA7219_HPF_MODE_MASK,
+ 0);
+ snd_soc_update_bits(codec, DA7219_DAC_FILTERS4, DA7219_DAC_EQ_EN_MASK,
+ 0);
+ snd_soc_update_bits(codec, DA7219_DAC_FILTERS5,
+ DA7219_DAC_SOFTMUTE_EN_MASK, 0);
+
+ /* Enable HP left & right paths */
+ snd_soc_update_bits(codec, DA7219_CP_CTRL, DA7219_CP_EN_MASK,
+ DA7219_CP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DIG_ROUTING_DAC,
+ DA7219_DAC_L_SRC_MASK | DA7219_DAC_R_SRC_MASK,
+ DA7219_DAC_L_SRC_TONEGEN |
+ DA7219_DAC_R_SRC_TONEGEN);
+ snd_soc_update_bits(codec, DA7219_DAC_L_CTRL,
+ DA7219_DAC_L_EN_MASK | DA7219_DAC_L_MUTE_EN_MASK,
+ DA7219_DAC_L_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DAC_R_CTRL,
+ DA7219_DAC_R_EN_MASK | DA7219_DAC_R_MUTE_EN_MASK,
+ DA7219_DAC_R_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_L_SELECT,
+ DA7219_MIXOUT_L_MIX_SELECT_MASK,
+ DA7219_MIXOUT_L_MIX_SELECT_MASK);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_R_SELECT,
+ DA7219_MIXOUT_R_MIX_SELECT_MASK,
+ DA7219_MIXOUT_R_MIX_SELECT_MASK);
+ snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1L,
+ DA7219_OUTFILT_ST_1L_SRC_MASK,
+ DA7219_DMIX_ST_SRC_OUTFILT1L);
+ snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1R,
+ DA7219_OUTFILT_ST_1R_SRC_MASK,
+ DA7219_DMIX_ST_SRC_OUTFILT1R);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_L_CTRL,
+ DA7219_MIXOUT_L_AMP_EN_MASK,
+ DA7219_MIXOUT_L_AMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL,
+ DA7219_MIXOUT_R_AMP_EN_MASK,
+ DA7219_MIXOUT_R_AMP_EN_MASK);
+ snd_soc_write(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK);
+ snd_soc_write(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK);
+
+ /* Configure & start Tone Generator */
+ snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK);
+ tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ);
+ regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L,
+ &tonegen_freq_hptest, sizeof(tonegen_freq_hptest));
+ snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2,
+ DA7219_SWG_SEL_MASK | DA7219_TONE_GEN_GAIN_MASK,
+ DA7219_SWG_SEL_SRAMP |
+ DA7219_TONE_GEN_GAIN_MINUS_15DB);
+ snd_soc_write(codec, DA7219_TONE_GEN_CFG1, DA7219_START_STOPN_MASK);
+
+ msleep(DA7219_AAD_HPTEST_PERIOD);
+
+ /* Grab comparator reading */
+ accdet_cfg8 = snd_soc_read(codec, DA7219_ACCDET_CONFIG_8);
+ if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK)
+ report |= SND_JACK_HEADPHONE;
+ else
+ report |= SND_JACK_LINEOUT;
+
+ /* Stop tone generator */
+ snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0);
+
+ msleep(DA7219_AAD_HPTEST_PERIOD);
+
+ /* Restore original settings from cache */
+ regcache_mark_dirty(da7219->regmap);
+ regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL,
+ DA7219_HP_R_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL,
+ DA7219_MIXOUT_R_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L,
+ DA7219_DROUTING_ST_OUTFILT_1R);
+ regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_SELECT,
+ DA7219_MIXOUT_R_SELECT);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_L_CTRL,
+ DA7219_DAC_R_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_DIG_ROUTING_DAC,
+ DA7219_DIG_ROUTING_DAC);
+ regcache_sync_region(da7219->regmap, DA7219_CP_CTRL, DA7219_CP_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS5,
+ DA7219_DAC_FILTERS5);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS4,
+ DA7219_DAC_FILTERS1);
+ regcache_sync_region(da7219->regmap, DA7219_HP_L_GAIN,
+ DA7219_HP_R_GAIN);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_L_GAIN,
+ DA7219_DAC_R_GAIN);
+ regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_ON_PER,
+ DA7219_TONE_GEN_ON_PER);
+ regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_FREQ1_L,
+ DA7219_TONE_GEN_FREQ1_U);
+ regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_CFG1,
+ DA7219_TONE_GEN_CFG2);
+
+ regcache_cache_bypass(da7219->regmap, false);
+
+ /* Disable HPTest block */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8,
+ DA7219_HPTEST_EN_MASK, 0);
+
+ /* Drive Headphones/lineout */
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK,
+ DA7219_HP_L_AMP_OE_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK,
+ DA7219_HP_R_AMP_OE_MASK);
+
+ mutex_unlock(&da7219->lock);
+ snd_soc_dapm_mutex_unlock(dapm);
+
+ /*
+ * Only send report if jack hasn't been removed during process,
+ * otherwise it's invalid and we drop it.
+ */
+ if (da7219_aad->jack_inserted)
+ snd_soc_jack_report(da7219_aad->jack, report,
+ SND_JACK_HEADSET | SND_JACK_LINEOUT);
+}
+
+
+/*
+ * IRQ
+ */
+
+static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
+{
+ struct da7219_aad_priv *da7219_aad = data;
+ struct snd_soc_codec *codec = da7219_aad->codec;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 events[DA7219_AAD_IRQ_REG_MAX];
+ u8 statusa;
+ int i, report = 0, mask = 0;
+
+ /* Read current IRQ events */
+ regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
+ events, DA7219_AAD_IRQ_REG_MAX);
+
+ if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B])
+ return IRQ_NONE;
+
+ /* Read status register for jack insertion & type status */
+ statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A);
+
+ /* Clear events */
+ regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
+ events, DA7219_AAD_IRQ_REG_MAX);
+
+ dev_dbg(codec->dev, "IRQ events = 0x%x|0x%x, status = 0x%x\n",
+ events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B],
+ statusa);
+
+ if (statusa & DA7219_JACK_INSERTION_STS_MASK) {
+ /* Jack Insertion */
+ if (events[DA7219_AAD_IRQ_REG_A] &
+ DA7219_E_JACK_INSERTED_MASK) {
+ report |= SND_JACK_MECHANICAL;
+ mask |= SND_JACK_MECHANICAL;
+ da7219_aad->jack_inserted = true;
+ }
+
+ /* Jack type detection */
+ if (events[DA7219_AAD_IRQ_REG_A] &
+ DA7219_E_JACK_DETECT_COMPLETE_MASK) {
+ /*
+ * If 4-pole, then enable button detection, else perform
+ * HP impedance test to determine output type to report.
+ *
+ * We schedule work here as the tasks themselves can
+ * take time to complete, and in particular for hptest
+ * we want to be able to check if the jack was removed
+ * during the procedure as this will invalidate the
+ * result. By doing this as work, the IRQ thread can
+ * handle a removal, and we can check at the end of
+ * hptest if we have a valid result or not.
+ */
+ if (statusa & DA7219_JACK_TYPE_STS_MASK) {
+ report |= SND_JACK_HEADSET;
+ mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT;
+ schedule_work(&da7219_aad->btn_det_work);
+ } else {
+ schedule_work(&da7219_aad->hptest_work);
+ }
+ }
+
+ /* Button support for 4-pole jack */
+ if (statusa & DA7219_JACK_TYPE_STS_MASK) {
+ for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) {
+ /* Button Press */
+ if (events[DA7219_AAD_IRQ_REG_B] &
+ (DA7219_E_BUTTON_A_PRESSED_MASK << i)) {
+ report |= SND_JACK_BTN_0 >> i;
+ mask |= SND_JACK_BTN_0 >> i;
+ }
+ }
+ snd_soc_jack_report(da7219_aad->jack, report, mask);
+
+ for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) {
+ /* Button Release */
+ if (events[DA7219_AAD_IRQ_REG_B] &
+ (DA7219_E_BUTTON_A_RELEASED_MASK >> i)) {
+ report &= ~(SND_JACK_BTN_0 >> i);
+ mask |= SND_JACK_BTN_0 >> i;
+ }
+ }
+ }
+ } else {
+ /* Jack removal */
+ if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_REMOVED_MASK) {
+ report = 0;
+ mask |= DA7219_AAD_REPORT_ALL_MASK;
+ da7219_aad->jack_inserted = false;
+
+ /* Un-drive headphones/lineout */
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_OE_MASK, 0);
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_OE_MASK, 0);
+
+ /* Ensure button detection disabled */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_BUTTON_CONFIG_MASK, 0);
+
+ /* Disable mic bias */
+ snd_soc_dapm_disable_pin(dapm, "Mic Bias");
+ snd_soc_dapm_sync(dapm);
+
+ /* Cancel any pending work */
+ cancel_work_sync(&da7219_aad->btn_det_work);
+ cancel_work_sync(&da7219_aad->hptest_work);
+ }
+ }
+
+ snd_soc_jack_report(da7219_aad->jack, report, mask);
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * DT to pdata conversion
+ */
+
+static enum da7219_aad_micbias_pulse_lvl
+ da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 2800:
+ return DA7219_AAD_MICBIAS_PULSE_LVL_2_8V;
+ case 2900:
+ return DA7219_AAD_MICBIAS_PULSE_LVL_2_9V;
+ default:
+ dev_warn(codec->dev, "Invalid micbias pulse level");
+ return DA7219_AAD_MICBIAS_PULSE_LVL_OFF;
+ }
+}
+
+static enum da7219_aad_btn_cfg
+ da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 2:
+ return DA7219_AAD_BTN_CFG_2MS;
+ case 5:
+ return DA7219_AAD_BTN_CFG_5MS;
+ case 10:
+ return DA7219_AAD_BTN_CFG_10MS;
+ case 50:
+ return DA7219_AAD_BTN_CFG_50MS;
+ case 100:
+ return DA7219_AAD_BTN_CFG_100MS;
+ case 200:
+ return DA7219_AAD_BTN_CFG_200MS;
+ case 500:
+ return DA7219_AAD_BTN_CFG_500MS;
+ default:
+ dev_warn(codec->dev, "Invalid button config");
+ return DA7219_AAD_BTN_CFG_10MS;
+ }
+}
+
+static enum da7219_aad_mic_det_thr
+ da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 200:
+ return DA7219_AAD_MIC_DET_THR_200_OHMS;
+ case 500:
+ return DA7219_AAD_MIC_DET_THR_500_OHMS;
+ case 750:
+ return DA7219_AAD_MIC_DET_THR_750_OHMS;
+ case 1000:
+ return DA7219_AAD_MIC_DET_THR_1000_OHMS;
+ default:
+ dev_warn(codec->dev, "Invalid mic detect threshold");
+ return DA7219_AAD_MIC_DET_THR_500_OHMS;
+ }
+}
+
+static enum da7219_aad_jack_ins_deb
+ da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 5:
+ return DA7219_AAD_JACK_INS_DEB_5MS;
+ case 10:
+ return DA7219_AAD_JACK_INS_DEB_10MS;
+ case 20:
+ return DA7219_AAD_JACK_INS_DEB_20MS;
+ case 50:
+ return DA7219_AAD_JACK_INS_DEB_50MS;
+ case 100:
+ return DA7219_AAD_JACK_INS_DEB_100MS;
+ case 200:
+ return DA7219_AAD_JACK_INS_DEB_200MS;
+ case 500:
+ return DA7219_AAD_JACK_INS_DEB_500MS;
+ case 1000:
+ return DA7219_AAD_JACK_INS_DEB_1S;
+ default:
+ dev_warn(codec->dev, "Invalid jack insert debounce");
+ return DA7219_AAD_JACK_INS_DEB_20MS;
+ }
+}
+
+static enum da7219_aad_jack_det_rate
+ da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "32ms_64ms")) {
+ return DA7219_AAD_JACK_DET_RATE_32_64MS;
+ } else if (!strcmp(str, "64ms_128ms")) {
+ return DA7219_AAD_JACK_DET_RATE_64_128MS;
+ } else if (!strcmp(str, "128ms_256ms")) {
+ return DA7219_AAD_JACK_DET_RATE_128_256MS;
+ } else if (!strcmp(str, "256ms_512ms")) {
+ return DA7219_AAD_JACK_DET_RATE_256_512MS;
+ } else {
+ dev_warn(codec->dev, "Invalid jack detect rate");
+ return DA7219_AAD_JACK_DET_RATE_256_512MS;
+ }
+}
+
+static enum da7219_aad_jack_rem_deb
+ da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1:
+ return DA7219_AAD_JACK_REM_DEB_1MS;
+ case 5:
+ return DA7219_AAD_JACK_REM_DEB_5MS;
+ case 10:
+ return DA7219_AAD_JACK_REM_DEB_10MS;
+ case 20:
+ return DA7219_AAD_JACK_REM_DEB_20MS;
+ default:
+ dev_warn(codec->dev, "Invalid jack removal debounce");
+ return DA7219_AAD_JACK_REM_DEB_1MS;
+ }
+}
+
+static enum da7219_aad_btn_avg
+ da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1:
+ return DA7219_AAD_BTN_AVG_1;
+ case 2:
+ return DA7219_AAD_BTN_AVG_2;
+ case 4:
+ return DA7219_AAD_BTN_AVG_4;
+ case 8:
+ return DA7219_AAD_BTN_AVG_8;
+ default:
+ dev_warn(codec->dev, "Invalid button average value");
+ return DA7219_AAD_BTN_AVG_2;
+ }
+}
+
+static enum da7219_aad_adc_1bit_rpt
+ da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1:
+ return DA7219_AAD_ADC_1BIT_RPT_1;
+ case 2:
+ return DA7219_AAD_ADC_1BIT_RPT_2;
+ case 4:
+ return DA7219_AAD_ADC_1BIT_RPT_4;
+ case 8:
+ return DA7219_AAD_ADC_1BIT_RPT_8;
+ default:
+ dev_warn(codec->dev, "Invalid ADC 1-bit repeat value");
+ return DA7219_AAD_ADC_1BIT_RPT_1;
+ }
+}
+
+static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec)
+{
+ struct device_node *np = codec->dev->of_node;
+ struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad");
+ struct da7219_aad_pdata *aad_pdata;
+ const char *of_str;
+ u32 of_val32;
+
+ if (!aad_np)
+ return NULL;
+
+ aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL);
+ if (!aad_pdata)
+ goto out;
+
+ aad_pdata->irq = irq_of_parse_and_map(np, 0);
+
+ if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl",
+ &of_val32) >= 0)
+ aad_pdata->micbias_pulse_lvl =
+ da7219_aad_of_micbias_pulse_lvl(codec, of_val32);
+ else
+ aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF;
+
+ if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time",
+ &of_val32) >= 0)
+ aad_pdata->micbias_pulse_time = of_val32;
+
+ if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0)
+ aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32);
+ else
+ aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS;
+
+ if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0)
+ aad_pdata->mic_det_thr =
+ da7219_aad_of_mic_det_thr(codec, of_val32);
+ else
+ aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS;
+
+ if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0)
+ aad_pdata->jack_ins_deb =
+ da7219_aad_of_jack_ins_deb(codec, of_val32);
+ else
+ aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS;
+
+ if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str))
+ aad_pdata->jack_det_rate =
+ da7219_aad_of_jack_det_rate(codec, of_str);
+ else
+ aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS;
+
+ if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0)
+ aad_pdata->jack_rem_deb =
+ da7219_aad_of_jack_rem_deb(codec, of_val32);
+ else
+ aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS;
+
+ if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0)
+ aad_pdata->a_d_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->a_d_btn_thr = 0xA;
+
+ if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0)
+ aad_pdata->d_b_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->d_b_btn_thr = 0x16;
+
+ if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0)
+ aad_pdata->b_c_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->b_c_btn_thr = 0x21;
+
+ if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0)
+ aad_pdata->c_mic_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->c_mic_btn_thr = 0x3E;
+
+ if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0)
+ aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32);
+ else
+ aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2;
+
+ if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0)
+ aad_pdata->adc_1bit_rpt =
+ da7219_aad_of_adc_1bit_rpt(codec, of_val32);
+ else
+ aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1;
+
+out:
+ of_node_put(aad_np);
+
+ return aad_pdata;
+}
+
+static void da7219_aad_handle_pdata(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_aad_priv *da7219_aad = da7219->aad;
+ struct da7219_pdata *pdata = da7219->pdata;
+
+ if ((pdata) && (pdata->aad_pdata)) {
+ struct da7219_aad_pdata *aad_pdata = pdata->aad_pdata;
+ u8 cfg, mask;
+
+ da7219_aad->irq = aad_pdata->irq;
+
+ switch (aad_pdata->micbias_pulse_lvl) {
+ case DA7219_AAD_MICBIAS_PULSE_LVL_2_8V:
+ case DA7219_AAD_MICBIAS_PULSE_LVL_2_9V:
+ da7219_aad->micbias_pulse_lvl =
+ (aad_pdata->micbias_pulse_lvl <<
+ DA7219_MICBIAS1_LEVEL_SHIFT);
+ break;
+ default:
+ break;
+ }
+
+ da7219_aad->micbias_pulse_time = aad_pdata->micbias_pulse_time;
+
+ switch (aad_pdata->btn_cfg) {
+ case DA7219_AAD_BTN_CFG_2MS:
+ case DA7219_AAD_BTN_CFG_5MS:
+ case DA7219_AAD_BTN_CFG_10MS:
+ case DA7219_AAD_BTN_CFG_50MS:
+ case DA7219_AAD_BTN_CFG_100MS:
+ case DA7219_AAD_BTN_CFG_200MS:
+ case DA7219_AAD_BTN_CFG_500MS:
+ da7219_aad->btn_cfg = (aad_pdata->btn_cfg <<
+ DA7219_BUTTON_CONFIG_SHIFT);
+ }
+
+ cfg = 0;
+ mask = 0;
+ switch (aad_pdata->mic_det_thr) {
+ case DA7219_AAD_MIC_DET_THR_200_OHMS:
+ case DA7219_AAD_MIC_DET_THR_500_OHMS:
+ case DA7219_AAD_MIC_DET_THR_750_OHMS:
+ case DA7219_AAD_MIC_DET_THR_1000_OHMS:
+ cfg |= (aad_pdata->mic_det_thr <<
+ DA7219_MIC_DET_THRESH_SHIFT);
+ mask |= DA7219_MIC_DET_THRESH_MASK;
+ }
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, mask, cfg);
+
+ cfg = 0;
+ mask = 0;
+ switch (aad_pdata->jack_ins_deb) {
+ case DA7219_AAD_JACK_INS_DEB_5MS:
+ case DA7219_AAD_JACK_INS_DEB_10MS:
+ case DA7219_AAD_JACK_INS_DEB_20MS:
+ case DA7219_AAD_JACK_INS_DEB_50MS:
+ case DA7219_AAD_JACK_INS_DEB_100MS:
+ case DA7219_AAD_JACK_INS_DEB_200MS:
+ case DA7219_AAD_JACK_INS_DEB_500MS:
+ case DA7219_AAD_JACK_INS_DEB_1S:
+ cfg |= (aad_pdata->jack_ins_deb <<
+ DA7219_JACKDET_DEBOUNCE_SHIFT);
+ mask |= DA7219_JACKDET_DEBOUNCE_MASK;
+ }
+ switch (aad_pdata->jack_det_rate) {
+ case DA7219_AAD_JACK_DET_RATE_32_64MS:
+ case DA7219_AAD_JACK_DET_RATE_64_128MS:
+ case DA7219_AAD_JACK_DET_RATE_128_256MS:
+ case DA7219_AAD_JACK_DET_RATE_256_512MS:
+ cfg |= (aad_pdata->jack_det_rate <<
+ DA7219_JACK_DETECT_RATE_SHIFT);
+ mask |= DA7219_JACK_DETECT_RATE_MASK;
+ }
+ switch (aad_pdata->jack_rem_deb) {
+ case DA7219_AAD_JACK_REM_DEB_1MS:
+ case DA7219_AAD_JACK_REM_DEB_5MS:
+ case DA7219_AAD_JACK_REM_DEB_10MS:
+ case DA7219_AAD_JACK_REM_DEB_20MS:
+ cfg |= (aad_pdata->jack_rem_deb <<
+ DA7219_JACKDET_REM_DEB_SHIFT);
+ mask |= DA7219_JACKDET_REM_DEB_MASK;
+ }
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_2, mask, cfg);
+
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_3,
+ aad_pdata->a_d_btn_thr);
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_4,
+ aad_pdata->d_b_btn_thr);
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_5,
+ aad_pdata->b_c_btn_thr);
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_6,
+ aad_pdata->c_mic_btn_thr);
+
+ cfg = 0;
+ mask = 0;
+ switch (aad_pdata->btn_avg) {
+ case DA7219_AAD_BTN_AVG_1:
+ case DA7219_AAD_BTN_AVG_2:
+ case DA7219_AAD_BTN_AVG_4:
+ case DA7219_AAD_BTN_AVG_8:
+ cfg |= (aad_pdata->btn_avg <<
+ DA7219_BUTTON_AVERAGE_SHIFT);
+ mask |= DA7219_BUTTON_AVERAGE_MASK;
+ }
+ switch (aad_pdata->adc_1bit_rpt) {
+ case DA7219_AAD_ADC_1BIT_RPT_1:
+ case DA7219_AAD_ADC_1BIT_RPT_2:
+ case DA7219_AAD_ADC_1BIT_RPT_4:
+ case DA7219_AAD_ADC_1BIT_RPT_8:
+ cfg |= (aad_pdata->adc_1bit_rpt <<
+ DA7219_ADC_1_BIT_REPEAT_SHIFT);
+ mask |= DA7219_ADC_1_BIT_REPEAT_MASK;
+ }
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_7, mask, cfg);
+ }
+}
+
+
+/*
+ * Init/Exit
+ */
+
+int da7219_aad_init(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_aad_priv *da7219_aad;
+ u8 mask[DA7219_AAD_IRQ_REG_MAX];
+ int ret;
+
+ da7219_aad = devm_kzalloc(codec->dev, sizeof(*da7219_aad), GFP_KERNEL);
+ if (!da7219_aad)
+ return -ENOMEM;
+
+ da7219->aad = da7219_aad;
+ da7219_aad->codec = codec;
+
+ /* Handle any DT/platform data */
+ if ((codec->dev->of_node) && (da7219->pdata))
+ da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec);
+
+ da7219_aad_handle_pdata(codec);
+
+ /* Disable button detection */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_BUTTON_CONFIG_MASK, 0);
+
+ INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work);
+ INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work);
+
+ ret = request_threaded_irq(da7219_aad->irq, NULL,
+ da7219_aad_irq_thread,
+ IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+ "da7219-aad", da7219_aad);
+ if (ret) {
+ dev_err(codec->dev, "Failed to request IRQ: %d\n", ret);
+ return ret;
+ }
+
+ /* Unmask AAD IRQs */
+ memset(mask, 0, DA7219_AAD_IRQ_REG_MAX);
+ regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A,
+ &mask, DA7219_AAD_IRQ_REG_MAX);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(da7219_aad_init);
+
+void da7219_aad_exit(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_aad_priv *da7219_aad = da7219->aad;
+ u8 mask[DA7219_AAD_IRQ_REG_MAX];
+
+ /* Mask off AAD IRQs */
+ memset(mask, DA7219_BYTE_MASK, DA7219_AAD_IRQ_REG_MAX);
+ regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A,
+ mask, DA7219_AAD_IRQ_REG_MAX);
+
+ free_irq(da7219_aad->irq, da7219_aad);
+
+ cancel_work_sync(&da7219_aad->btn_det_work);
+ cancel_work_sync(&da7219_aad->hptest_work);
+}
+EXPORT_SYMBOL_GPL(da7219_aad_exit);
+
+MODULE_DESCRIPTION("ASoC DA7219 AAD Driver");
+MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h
new file mode 100644
index 0000000..4fccf67
--- /dev/null
+++ b/sound/soc/codecs/da7219-aad.h
@@ -0,0 +1,212 @@
+/*
+ * da7219-aad.h - DA7322 ASoC AAD Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor Ltd.
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_AAD_H
+#define __DA7219_AAD_H
+
+#include <linux/timer.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/da7219-aad.h>
+
+/*
+ * Registers
+ */
+
+#define DA7219_ACCDET_STATUS_A 0xC0
+#define DA7219_ACCDET_STATUS_B 0xC1
+#define DA7219_ACCDET_IRQ_EVENT_A 0xC2
+#define DA7219_ACCDET_IRQ_EVENT_B 0xC3
+#define DA7219_ACCDET_IRQ_MASK_A 0xC4
+#define DA7219_ACCDET_IRQ_MASK_B 0xC5
+#define DA7219_ACCDET_CONFIG_1 0xC6
+#define DA7219_ACCDET_CONFIG_2 0xC7
+#define DA7219_ACCDET_CONFIG_3 0xC8
+#define DA7219_ACCDET_CONFIG_4 0xC9
+#define DA7219_ACCDET_CONFIG_5 0xCA
+#define DA7219_ACCDET_CONFIG_6 0xCB
+#define DA7219_ACCDET_CONFIG_7 0xCC
+#define DA7219_ACCDET_CONFIG_8 0xCD
+
+
+/*
+ * Bit Fields
+ */
+
+/* DA7219_ACCDET_STATUS_A = 0xC0 */
+#define DA7219_JACK_INSERTION_STS_SHIFT 0
+#define DA7219_JACK_INSERTION_STS_MASK (0x1 << 0)
+#define DA7219_JACK_TYPE_STS_SHIFT 1
+#define DA7219_JACK_TYPE_STS_MASK (0x1 << 1)
+#define DA7219_JACK_PIN_ORDER_STS_SHIFT 2
+#define DA7219_JACK_PIN_ORDER_STS_MASK (0x1 << 2)
+#define DA7219_MICBIAS_UP_STS_SHIFT 3
+#define DA7219_MICBIAS_UP_STS_MASK (0x1 << 3)
+
+/* DA7219_ACCDET_STATUS_B = 0xC1 */
+#define DA7219_BUTTON_TYPE_STS_SHIFT 0
+#define DA7219_BUTTON_TYPE_STS_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_IRQ_EVENT_A = 0xC2 */
+#define DA7219_E_JACK_INSERTED_SHIFT 0
+#define DA7219_E_JACK_INSERTED_MASK (0x1 << 0)
+#define DA7219_E_JACK_REMOVED_SHIFT 1
+#define DA7219_E_JACK_REMOVED_MASK (0x1 << 1)
+#define DA7219_E_JACK_DETECT_COMPLETE_SHIFT 2
+#define DA7219_E_JACK_DETECT_COMPLETE_MASK (0x1 << 2)
+
+/* DA7219_ACCDET_IRQ_EVENT_B = 0xC3 */
+#define DA7219_E_BUTTON_A_PRESSED_SHIFT 0
+#define DA7219_E_BUTTON_A_PRESSED_MASK (0x1 << 0)
+#define DA7219_E_BUTTON_B_PRESSED_SHIFT 1
+#define DA7219_E_BUTTON_B_PRESSED_MASK (0x1 << 1)
+#define DA7219_E_BUTTON_C_PRESSED_SHIFT 2
+#define DA7219_E_BUTTON_C_PRESSED_MASK (0x1 << 2)
+#define DA7219_E_BUTTON_D_PRESSED_SHIFT 3
+#define DA7219_E_BUTTON_D_PRESSED_MASK (0x1 << 3)
+#define DA7219_E_BUTTON_D_RELEASED_SHIFT 4
+#define DA7219_E_BUTTON_D_RELEASED_MASK (0x1 << 4)
+#define DA7219_E_BUTTON_C_RELEASED_SHIFT 5
+#define DA7219_E_BUTTON_C_RELEASED_MASK (0x1 << 5)
+#define DA7219_E_BUTTON_B_RELEASED_SHIFT 6
+#define DA7219_E_BUTTON_B_RELEASED_MASK (0x1 << 6)
+#define DA7219_E_BUTTON_A_RELEASED_SHIFT 7
+#define DA7219_E_BUTTON_A_RELEASED_MASK (0x1 << 7)
+
+/* DA7219_ACCDET_IRQ_MASK_A = 0xC4 */
+#define DA7219_M_JACK_INSERTED_SHIFT 0
+#define DA7219_M_JACK_INSERTED_MASK (0x1 << 0)
+#define DA7219_M_JACK_REMOVED_SHIFT 1
+#define DA7219_M_JACK_REMOVED_MASK (0x1 << 1)
+#define DA7219_M_JACK_DETECT_COMPLETE_SHIFT 2
+#define DA7219_M_JACK_DETECT_COMPLETE_MASK (0x1 << 2)
+
+/* DA7219_ACCDET_IRQ_MASK_B = 0xC5 */
+#define DA7219_M_BUTTON_A_PRESSED_SHIFT 0
+#define DA7219_M_BUTTON_A_PRESSED_MASK (0x1 << 0)
+#define DA7219_M_BUTTON_B_PRESSED_SHIFT 1
+#define DA7219_M_BUTTON_B_PRESSED_MASK (0x1 << 1)
+#define DA7219_M_BUTTON_C_PRESSED_SHIFT 2
+#define DA7219_M_BUTTON_C_PRESSED_MASK (0x1 << 2)
+#define DA7219_M_BUTTON_D_PRESSED_SHIFT 3
+#define DA7219_M_BUTTON_D_PRESSED_MASK (0x1 << 3)
+#define DA7219_M_BUTTON_D_RELEASED_SHIFT 4
+#define DA7219_M_BUTTON_D_RELEASED_MASK (0x1 << 4)
+#define DA7219_M_BUTTON_C_RELEASED_SHIFT 5
+#define DA7219_M_BUTTON_C_RELEASED_MASK (0x1 << 5)
+#define DA7219_M_BUTTON_B_RELEASED_SHIFT 6
+#define DA7219_M_BUTTON_B_RELEASED_MASK (0x1 << 6)
+#define DA7219_M_BUTTON_A_RELEASED_SHIFT 7
+#define DA7219_M_BUTTON_A_RELEASED_MASK (0x1 << 7)
+
+/* DA7219_ACCDET_CONFIG_1 = 0xC6 */
+#define DA7219_ACCDET_EN_SHIFT 0
+#define DA7219_ACCDET_EN_MASK (0x1 << 0)
+#define DA7219_BUTTON_CONFIG_SHIFT 1
+#define DA7219_BUTTON_CONFIG_MASK (0x7 << 1)
+#define DA7219_MIC_DET_THRESH_SHIFT 4
+#define DA7219_MIC_DET_THRESH_MASK (0x3 << 4)
+#define DA7219_JACK_TYPE_DET_EN_SHIFT 6
+#define DA7219_JACK_TYPE_DET_EN_MASK (0x1 << 6)
+#define DA7219_PIN_ORDER_DET_EN_SHIFT 7
+#define DA7219_PIN_ORDER_DET_EN_MASK (0x1 << 7)
+
+/* DA7219_ACCDET_CONFIG_2 = 0xC7 */
+#define DA7219_ACCDET_PAUSE_SHIFT 0
+#define DA7219_ACCDET_PAUSE_MASK (0x1 << 0)
+#define DA7219_JACKDET_DEBOUNCE_SHIFT 1
+#define DA7219_JACKDET_DEBOUNCE_MASK (0x7 << 1)
+#define DA7219_JACK_DETECT_RATE_SHIFT 4
+#define DA7219_JACK_DETECT_RATE_MASK (0x3 << 4)
+#define DA7219_JACKDET_REM_DEB_SHIFT 6
+#define DA7219_JACKDET_REM_DEB_MASK (0x3 << 6)
+
+/* DA7219_ACCDET_CONFIG_3 = 0xC8 */
+#define DA7219_A_D_BUTTON_THRESH_SHIFT 0
+#define DA7219_A_D_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_4 = 0xC9 */
+#define DA7219_D_B_BUTTON_THRESH_SHIFT 0
+#define DA7219_D_B_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_5 = 0xCA */
+#define DA7219_B_C_BUTTON_THRESH_SHIFT 0
+#define DA7219_B_C_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_6 = 0xCB */
+#define DA7219_C_MIC_BUTTON_THRESH_SHIFT 0
+#define DA7219_C_MIC_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_7 = 0xCC */
+#define DA7219_BUTTON_AVERAGE_SHIFT 0
+#define DA7219_BUTTON_AVERAGE_MASK (0x3 << 0)
+#define DA7219_ADC_1_BIT_REPEAT_SHIFT 2
+#define DA7219_ADC_1_BIT_REPEAT_MASK (0x3 << 2)
+#define DA7219_PIN_ORDER_FORCE_SHIFT 4
+#define DA7219_PIN_ORDER_FORCE_MASK (0x1 << 4)
+#define DA7219_JACK_TYPE_FORCE_SHIFT 5
+#define DA7219_JACK_TYPE_FORCE_MASK (0x1 << 5)
+
+/* DA7219_ACCDET_CONFIG_8 = 0xCD */
+#define DA7219_HPTEST_EN_SHIFT 0
+#define DA7219_HPTEST_EN_MASK (0x1 << 0)
+#define DA7219_HPTEST_RES_SEL_SHIFT 1
+#define DA7219_HPTEST_RES_SEL_MASK (0x3 << 1)
+#define DA7219_HPTEST_RES_SEL_1KOHMS (0x0 << 1)
+#define DA7219_HPTEST_COMP_SHIFT 4
+#define DA7219_HPTEST_COMP_MASK (0x1 << 4)
+
+
+#define DA7219_AAD_MAX_BUTTONS 4
+#define DA7219_AAD_REPORT_ALL_MASK (SND_JACK_MECHANICAL | \
+ SND_JACK_HEADSET | SND_JACK_LINEOUT | \
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 | \
+ SND_JACK_BTN_2 | SND_JACK_BTN_3)
+
+#define DA7219_AAD_MICBIAS_CHK_DELAY 10
+#define DA7219_AAD_MICBIAS_CHK_RETRIES 5
+
+#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28
+#define DA7219_AAD_HPTEST_PERIOD 65
+
+enum da7219_aad_event_regs {
+ DA7219_AAD_IRQ_REG_A = 0,
+ DA7219_AAD_IRQ_REG_B,
+ DA7219_AAD_IRQ_REG_MAX,
+};
+
+/* Private data */
+struct da7219_aad_priv {
+ struct snd_soc_codec *codec;
+ int irq;
+
+ u8 micbias_pulse_lvl;
+ u32 micbias_pulse_time;
+
+ u8 btn_cfg;
+
+ struct work_struct btn_det_work;
+ struct work_struct hptest_work;
+
+ struct snd_soc_jack *jack;
+ bool jack_inserted;
+};
+
+/* AAD control */
+void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
+
+/* Init/Exit */
+int da7219_aad_init(struct snd_soc_codec *codec);
+void da7219_aad_exit(struct snd_soc_codec *codec);
+
+#endif /* __DA7219_AAD_H */
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
new file mode 100644
index 0000000..f238c1e
--- /dev/null
+++ b/sound/soc/codecs/da7219.c
@@ -0,0 +1,1955 @@
+/*
+ * da7219.c - DA7219 ALSA SoC Codec Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/of_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/pm.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/regulator/consumer.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include <sound/da7219.h>
+#include "da7219.h"
+#include "da7219-aad.h"
+
+
+/*
+ * TLVs and Enums
+ */
+
+/* Input TLVs */
+static const DECLARE_TLV_DB_SCALE(da7219_mic_gain_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_mixin_gain_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_adc_dig_gain_tlv, -8325, 75, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_threshold_tlv, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_gain_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_ana_gain_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_sidetone_gain_tlv, -4200, 300, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_tonegen_gain_tlv, -4500, 300, 0);
+
+/* Output TLVs */
+static const DECLARE_TLV_DB_SCALE(da7219_dac_eq_band_tlv, -1050, 150, 0);
+
+static const DECLARE_TLV_DB_RANGE(da7219_dac_dig_gain_tlv,
+ 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+ /* -77.25dB to 12dB */
+ 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7725, 75, 0)
+);
+
+static const DECLARE_TLV_DB_SCALE(da7219_dac_ng_threshold_tlv, -10200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_hp_gain_tlv, -5700, 100, 0);
+
+/* Input Enums */
+static const char * const da7219_alc_attack_rate_txt[] = {
+ "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs",
+ "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs",
+ "30024/fs"
+};
+
+static const struct soc_enum da7219_alc_attack_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_ATTACK_SHIFT,
+ DA7219_ALC_ATTACK_MAX, da7219_alc_attack_rate_txt);
+
+static const char * const da7219_alc_release_rate_txt[] = {
+ "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs",
+ "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs"
+};
+
+static const struct soc_enum da7219_alc_release_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_RELEASE_SHIFT,
+ DA7219_ALC_RELEASE_MAX, da7219_alc_release_rate_txt);
+
+static const char * const da7219_alc_hold_time_txt[] = {
+ "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs",
+ "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs",
+ "253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
+};
+
+static const struct soc_enum da7219_alc_hold_time =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_HOLD_SHIFT,
+ DA7219_ALC_HOLD_MAX, da7219_alc_hold_time_txt);
+
+static const char * const da7219_alc_env_rate_txt[] = {
+ "1/4", "1/16", "1/256", "1/65536"
+};
+
+static const struct soc_enum da7219_alc_env_attack_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_ATTACK_SHIFT,
+ DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt);
+
+static const struct soc_enum da7219_alc_env_release_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_RELEASE_SHIFT,
+ DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt);
+
+static const char * const da7219_alc_anticlip_step_txt[] = {
+ "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs"
+};
+
+static const struct soc_enum da7219_alc_anticlip_step =
+ SOC_ENUM_SINGLE(DA7219_ALC_ANTICLIP_CTRL,
+ DA7219_ALC_ANTICLIP_STEP_SHIFT,
+ DA7219_ALC_ANTICLIP_STEP_MAX,
+ da7219_alc_anticlip_step_txt);
+
+/* Input/Output Enums */
+static const char * const da7219_gain_ramp_rate_txt[] = {
+ "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8",
+ "Nominal Rate / 16"
+};
+
+static const struct soc_enum da7219_gain_ramp_rate =
+ SOC_ENUM_SINGLE(DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_SHIFT,
+ DA7219_GAIN_RAMP_RATE_MAX, da7219_gain_ramp_rate_txt);
+
+static const char * const da7219_hpf_mode_txt[] = {
+ "Disabled", "Audio", "Voice"
+};
+
+static const unsigned int da7219_hpf_mode_val[] = {
+ DA7219_HPF_DISABLED, DA7219_HPF_AUDIO_EN, DA7219_HPF_VOICE_EN,
+};
+
+static const struct soc_enum da7219_adc_hpf_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7219_ADC_FILTERS1, DA7219_HPF_MODE_SHIFT,
+ DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX,
+ da7219_hpf_mode_txt, da7219_hpf_mode_val);
+
+static const struct soc_enum da7219_dac_hpf_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7219_DAC_FILTERS1, DA7219_HPF_MODE_SHIFT,
+ DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX,
+ da7219_hpf_mode_txt, da7219_hpf_mode_val);
+
+static const char * const da7219_audio_hpf_corner_txt[] = {
+ "2Hz", "4Hz", "8Hz", "16Hz"
+};
+
+static const struct soc_enum da7219_adc_audio_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1,
+ DA7219_ADC_AUDIO_HPF_CORNER_SHIFT,
+ DA7219_AUDIO_HPF_CORNER_MAX,
+ da7219_audio_hpf_corner_txt);
+
+static const struct soc_enum da7219_dac_audio_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1,
+ DA7219_DAC_AUDIO_HPF_CORNER_SHIFT,
+ DA7219_AUDIO_HPF_CORNER_MAX,
+ da7219_audio_hpf_corner_txt);
+
+static const char * const da7219_voice_hpf_corner_txt[] = {
+ "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da7219_adc_voice_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1,
+ DA7219_ADC_VOICE_HPF_CORNER_SHIFT,
+ DA7219_VOICE_HPF_CORNER_MAX,
+ da7219_voice_hpf_corner_txt);
+
+static const struct soc_enum da7219_dac_voice_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1,
+ DA7219_DAC_VOICE_HPF_CORNER_SHIFT,
+ DA7219_VOICE_HPF_CORNER_MAX,
+ da7219_voice_hpf_corner_txt);
+
+static const char * const da7219_tonegen_dtmf_key_txt[] = {
+ "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D",
+ "*", "#"
+};
+
+static const struct soc_enum da7219_tonegen_dtmf_key =
+ SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG1, DA7219_DTMF_REG_SHIFT,
+ DA7219_DTMF_REG_MAX, da7219_tonegen_dtmf_key_txt);
+
+static const char * const da7219_tonegen_swg_sel_txt[] = {
+ "Sum", "SWG1", "SWG2", "SWG1_1-Cos"
+};
+
+static const struct soc_enum da7219_tonegen_swg_sel =
+ SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG2, DA7219_SWG_SEL_SHIFT,
+ DA7219_SWG_SEL_MAX, da7219_tonegen_swg_sel_txt);
+
+/* Output Enums */
+static const char * const da7219_dac_softmute_rate_txt[] = {
+ "1 Sample", "2 Samples", "4 Samples", "8 Samples", "16 Samples",
+ "32 Samples", "64 Samples"
+};
+
+static const struct soc_enum da7219_dac_softmute_rate =
+ SOC_ENUM_SINGLE(DA7219_DAC_FILTERS5, DA7219_DAC_SOFTMUTE_RATE_SHIFT,
+ DA7219_DAC_SOFTMUTE_RATE_MAX,
+ da7219_dac_softmute_rate_txt);
+
+static const char * const da7219_dac_ng_setup_time_txt[] = {
+ "256 Samples", "512 Samples", "1024 Samples", "2048 Samples"
+};
+
+static const struct soc_enum da7219_dac_ng_setup_time =
+ SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+ DA7219_DAC_NG_SETUP_TIME_SHIFT,
+ DA7219_DAC_NG_SETUP_TIME_MAX,
+ da7219_dac_ng_setup_time_txt);
+
+static const char * const da7219_dac_ng_rampup_txt[] = {
+ "0.22ms/dB", "0.0138ms/dB"
+};
+
+static const struct soc_enum da7219_dac_ng_rampup_rate =
+ SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+ DA7219_DAC_NG_RAMPUP_RATE_SHIFT,
+ DA7219_DAC_NG_RAMP_RATE_MAX,
+ da7219_dac_ng_rampup_txt);
+
+static const char * const da7219_dac_ng_rampdown_txt[] = {
+ "0.88ms/dB", "14.08ms/dB"
+};
+
+static const struct soc_enum da7219_dac_ng_rampdown_rate =
+ SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+ DA7219_DAC_NG_RAMPDN_RATE_SHIFT,
+ DA7219_DAC_NG_RAMP_RATE_MAX,
+ da7219_dac_ng_rampdown_txt);
+
+
+static const char * const da7219_cp_track_mode_txt[] = {
+ "Largest Volume", "DAC Volume", "Signal Magnitude"
+};
+
+static const unsigned int da7219_cp_track_mode_val[] = {
+ DA7219_CP_MCHANGE_LARGEST_VOL, DA7219_CP_MCHANGE_DAC_VOL,
+ DA7219_CP_MCHANGE_SIG_MAG
+};
+
+static const struct soc_enum da7219_cp_track_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7219_CP_CTRL, DA7219_CP_MCHANGE_SHIFT,
+ DA7219_CP_MCHANGE_REL_MASK, DA7219_CP_MCHANGE_MAX,
+ da7219_cp_track_mode_txt,
+ da7219_cp_track_mode_val);
+
+
+/*
+ * Control Functions
+ */
+
+/* Locked Kcontrol calls */
+static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_get_volsw(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_get_enum_double(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_put_enum_double(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+/* ALC */
+static void da7219_alc_calib(struct snd_soc_codec *codec)
+{
+ u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl;
+
+ /* Save current state of mic control register */
+ mic_ctrl = snd_soc_read(codec, DA7219_MIC_1_CTRL);
+
+ /* Save current state of input mixer control register */
+ mixin_ctrl = snd_soc_read(codec, DA7219_MIXIN_L_CTRL);
+
+ /* Save current state of input ADC control register */
+ adc_ctrl = snd_soc_read(codec, DA7219_ADC_L_CTRL);
+
+ /* Enable then Mute MIC PGAs */
+ snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK,
+ DA7219_MIC_1_AMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_MIC_1_CTRL,
+ DA7219_MIC_1_AMP_MUTE_EN_MASK,
+ DA7219_MIC_1_AMP_MUTE_EN_MASK);
+
+ /* Enable input mixers unmuted */
+ snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_EN_MASK |
+ DA7219_MIXIN_L_AMP_MUTE_EN_MASK,
+ DA7219_MIXIN_L_AMP_EN_MASK);
+
+ /* Enable input filters unmuted */
+ snd_soc_update_bits(codec, DA7219_ADC_L_CTRL,
+ DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK,
+ DA7219_ADC_L_EN_MASK);
+
+ /* Perform auto calibration */
+ snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+ DA7219_ALC_AUTO_CALIB_EN_MASK,
+ DA7219_ALC_AUTO_CALIB_EN_MASK);
+ do {
+ calib_ctrl = snd_soc_read(codec, DA7219_ALC_CTRL1);
+ } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK);
+
+ /* If auto calibration fails, disable DC offset, hybrid ALC */
+ if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) {
+ dev_warn(codec->dev,
+ "ALC auto calibration failed with overflow\n");
+ snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+ DA7219_ALC_OFFSET_EN_MASK |
+ DA7219_ALC_SYNC_MODE_MASK, 0);
+ } else {
+ /* Enable DC offset cancellation, hybrid mode */
+ snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+ DA7219_ALC_OFFSET_EN_MASK |
+ DA7219_ALC_SYNC_MODE_MASK,
+ DA7219_ALC_OFFSET_EN_MASK |
+ DA7219_ALC_SYNC_MODE_MASK);
+ }
+
+ /* Restore input filter control register to original state */
+ snd_soc_write(codec, DA7219_ADC_L_CTRL, adc_ctrl);
+
+ /* Restore input mixer control registers to original state */
+ snd_soc_write(codec, DA7219_MIXIN_L_CTRL, mixin_ctrl);
+
+ /* Restore MIC control registers to original states */
+ snd_soc_write(codec, DA7219_MIC_1_CTRL, mic_ctrl);
+}
+
+static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+
+ /*
+ * If ALC in operation and value of control has been updated,
+ * make sure calibrated offsets are updated.
+ */
+ if ((ret == 1) && (da7219->alc_en))
+ da7219_alc_calib(codec);
+
+ return ret;
+}
+
+static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+
+ /* Force ALC offset calibration if enabling ALC */
+ if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) {
+ da7219_alc_calib(codec);
+ da7219->alc_en = true;
+ } else {
+ da7219->alc_en = false;
+ }
+
+ return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
+/* ToneGen */
+static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int reg = mixer_ctrl->reg;
+ u16 val;
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val));
+ mutex_unlock(&da7219->lock);
+
+ if (ret)
+ return ret;
+
+ /*
+ * Frequency value spans two 8-bit registers, lower then upper byte.
+ * Therefore we need to convert to host endianness here.
+ */
+ ucontrol->value.integer.value[0] = le16_to_cpu(val);
+
+ return 0;
+}
+
+static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int reg = mixer_ctrl->reg;
+ u16 val;
+ int ret;
+
+ /*
+ * Frequency value spans two 8-bit registers, lower then upper byte.
+ * Therefore we need to convert to little endian here to align with
+ * HW registers.
+ */
+ val = cpu_to_le16(ucontrol->value.integer.value[0]);
+
+ mutex_lock(&da7219->lock);
+ ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val));
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+
+/*
+ * KControls
+ */
+
+static const struct snd_kcontrol_new da7219_snd_controls[] = {
+ /* Mics */
+ SOC_SINGLE_TLV("Mic Volume", DA7219_MIC_1_GAIN,
+ DA7219_MIC_1_AMP_GAIN_SHIFT, DA7219_MIC_1_AMP_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_mic_gain_tlv),
+ SOC_SINGLE("Mic Switch", DA7219_MIC_1_CTRL,
+ DA7219_MIC_1_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+
+ /* Mixer Input */
+ SOC_SINGLE_EXT_TLV("Mixin Volume", DA7219_MIXIN_L_GAIN,
+ DA7219_MIXIN_L_AMP_GAIN_SHIFT,
+ DA7219_MIXIN_L_AMP_GAIN_MAX, DA7219_NO_INVERT,
+ snd_soc_get_volsw, da7219_mixin_gain_put,
+ da7219_mixin_gain_tlv),
+ SOC_SINGLE("Mixin Switch", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+ SOC_SINGLE("Mixin Gain Ramp Switch", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+ SOC_SINGLE("Mixin ZC Gain Switch", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_ZC_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+
+ /* ADC */
+ SOC_SINGLE_TLV("Capture Digital Volume", DA7219_ADC_L_GAIN,
+ DA7219_ADC_L_DIGITAL_GAIN_SHIFT,
+ DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT,
+ da7219_adc_dig_gain_tlv),
+ SOC_SINGLE("Capture Digital Switch", DA7219_ADC_L_CTRL,
+ DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+ SOC_SINGLE("Capture Digital Gain Ramp Switch", DA7219_ADC_L_CTRL,
+ DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+
+ /* ALC */
+ SOC_ENUM("ALC Attack Rate", da7219_alc_attack_rate),
+ SOC_ENUM("ALC Release Rate", da7219_alc_release_rate),
+ SOC_ENUM("ALC Hold Time", da7219_alc_hold_time),
+ SOC_ENUM("ALC Envelope Attack Rate", da7219_alc_env_attack_rate),
+ SOC_ENUM("ALC Envelope Release Rate", da7219_alc_env_release_rate),
+ SOC_SINGLE_TLV("ALC Noise Threshold", DA7219_ALC_NOISE,
+ DA7219_ALC_NOISE_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+ DA7219_INVERT, da7219_alc_threshold_tlv),
+ SOC_SINGLE_TLV("ALC Min Threshold", DA7219_ALC_TARGET_MIN,
+ DA7219_ALC_THRESHOLD_MIN_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+ DA7219_INVERT, da7219_alc_threshold_tlv),
+ SOC_SINGLE_TLV("ALC Max Threshold", DA7219_ALC_TARGET_MAX,
+ DA7219_ALC_THRESHOLD_MAX_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+ DA7219_INVERT, da7219_alc_threshold_tlv),
+ SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS,
+ DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_gain_tlv),
+ SOC_SINGLE_TLV("ALC Max Volume", DA7219_ALC_GAIN_LIMITS,
+ DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("ALC Min Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS,
+ DA7219_ALC_ANA_GAIN_MIN_SHIFT,
+ DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_ana_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("ALC Max Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS,
+ DA7219_ALC_ANA_GAIN_MAX_SHIFT,
+ DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_ana_gain_tlv),
+ SOC_ENUM("ALC Anticlip Step", da7219_alc_anticlip_step),
+ SOC_SINGLE("ALC Anticlip Switch", DA7219_ALC_ANTICLIP_CTRL,
+ DA7219_ALC_ANTIPCLIP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+ SOC_SINGLE_EXT("ALC Switch", DA7219_ALC_CTRL1, DA7219_ALC_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT,
+ snd_soc_get_volsw, da7219_alc_sw_put),
+
+ /* Input High-Pass Filters */
+ SOC_ENUM("ADC HPF Mode", da7219_adc_hpf_mode),
+ SOC_ENUM("ADC HPF Corner Audio", da7219_adc_audio_hpf_corner),
+ SOC_ENUM("ADC HPF Corner Voice", da7219_adc_voice_hpf_corner),
+
+ /* Sidetone Filter */
+ SOC_SINGLE_TLV("Sidetone Volume", DA7219_SIDETONE_GAIN,
+ DA7219_SIDETONE_GAIN_SHIFT, DA7219_SIDETONE_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_sidetone_gain_tlv),
+ SOC_SINGLE("Sidetone Switch", DA7219_SIDETONE_CTRL,
+ DA7219_SIDETONE_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+
+ /* Tone Generator */
+ SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7219_TONE_GEN_CFG2,
+ DA7219_TONE_GEN_GAIN_SHIFT, DA7219_TONE_GEN_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put, da7219_tonegen_gain_tlv),
+ SOC_ENUM_EXT("ToneGen DTMF Key", da7219_tonegen_dtmf_key,
+ da7219_enum_locked_get, da7219_enum_locked_put),
+ SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7219_TONE_GEN_CFG1,
+ DA7219_DTMF_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+ SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7219_tonegen_swg_sel,
+ da7219_enum_locked_get, da7219_enum_locked_put),
+ SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7219_TONE_GEN_FREQ1_L,
+ DA7219_FREQ1_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT,
+ da7219_tonegen_freq_get, da7219_tonegen_freq_put),
+ SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7219_TONE_GEN_FREQ2_L,
+ DA7219_FREQ2_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT,
+ da7219_tonegen_freq_get, da7219_tonegen_freq_put),
+ SOC_SINGLE_EXT("ToneGen On Time", DA7219_TONE_GEN_ON_PER,
+ DA7219_BEEP_ON_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+ SOC_SINGLE("ToneGen Off Time", DA7219_TONE_GEN_OFF_PER,
+ DA7219_BEEP_OFF_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX,
+ DA7219_NO_INVERT),
+
+ /* Gain ramping */
+ SOC_ENUM("Gain Ramp Rate", da7219_gain_ramp_rate),
+
+ /* DAC High-Pass Filter */
+ SOC_ENUM_EXT("DAC HPF Mode", da7219_dac_hpf_mode,
+ da7219_enum_locked_get, da7219_enum_locked_put),
+ SOC_ENUM("DAC HPF Corner Audio", da7219_dac_audio_hpf_corner),
+ SOC_ENUM("DAC HPF Corner Voice", da7219_dac_voice_hpf_corner),
+
+ /* DAC 5-Band Equaliser */
+ SOC_SINGLE_TLV("DAC EQ Band1 Volume", DA7219_DAC_FILTERS2,
+ DA7219_DAC_EQ_BAND1_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band2 Volume", DA7219_DAC_FILTERS2,
+ DA7219_DAC_EQ_BAND2_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band3 Volume", DA7219_DAC_FILTERS3,
+ DA7219_DAC_EQ_BAND3_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band4 Volume", DA7219_DAC_FILTERS3,
+ DA7219_DAC_EQ_BAND4_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band5 Volume", DA7219_DAC_FILTERS4,
+ DA7219_DAC_EQ_BAND5_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_EXT("DAC EQ Switch", DA7219_DAC_FILTERS4,
+ DA7219_DAC_EQ_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+
+ /* DAC Softmute */
+ SOC_ENUM("DAC Soft Mute Rate", da7219_dac_softmute_rate),
+ SOC_SINGLE_EXT("DAC Soft Mute Switch", DA7219_DAC_FILTERS5,
+ DA7219_DAC_SOFTMUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+
+ /* DAC Noise Gate */
+ SOC_ENUM("DAC NG Setup Time", da7219_dac_ng_setup_time),
+ SOC_ENUM("DAC NG Rampup Rate", da7219_dac_ng_rampup_rate),
+ SOC_ENUM("DAC NG Rampdown Rate", da7219_dac_ng_rampdown_rate),
+ SOC_SINGLE_TLV("DAC NG Off Threshold", DA7219_DAC_NG_OFF_THRESH,
+ DA7219_DAC_NG_OFF_THRESHOLD_SHIFT,
+ DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT,
+ da7219_dac_ng_threshold_tlv),
+ SOC_SINGLE_TLV("DAC NG On Threshold", DA7219_DAC_NG_ON_THRESH,
+ DA7219_DAC_NG_ON_THRESHOLD_SHIFT,
+ DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT,
+ da7219_dac_ng_threshold_tlv),
+ SOC_SINGLE("DAC NG Switch", DA7219_DAC_NG_CTRL, DA7219_DAC_NG_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+
+ /* DACs */
+ SOC_DOUBLE_R_EXT_TLV("Playback Digital Volume", DA7219_DAC_L_GAIN,
+ DA7219_DAC_R_GAIN, DA7219_DAC_L_DIGITAL_GAIN_SHIFT,
+ DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT,
+ da7219_volsw_locked_get, da7219_volsw_locked_put,
+ da7219_dac_dig_gain_tlv),
+ SOC_DOUBLE_R_EXT("Playback Digital Switch", DA7219_DAC_L_CTRL,
+ DA7219_DAC_R_CTRL, DA7219_DAC_L_MUTE_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_INVERT,
+ da7219_volsw_locked_get, da7219_volsw_locked_put),
+ SOC_DOUBLE_R("Playback Digital Gain Ramp Switch", DA7219_DAC_L_CTRL,
+ DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+
+ /* CP */
+ SOC_ENUM("Charge Pump Track Mode", da7219_cp_track_mode),
+ SOC_SINGLE("Charge Pump Threshold", DA7219_CP_VOL_THRESHOLD1,
+ DA7219_CP_THRESH_VDD2_SHIFT, DA7219_CP_THRESH_VDD2_MAX,
+ DA7219_NO_INVERT),
+
+ /* Headphones */
+ SOC_DOUBLE_R_EXT_TLV("Headphone Volume", DA7219_HP_L_GAIN,
+ DA7219_HP_R_GAIN, DA7219_HP_L_AMP_GAIN_SHIFT,
+ DA7219_HP_AMP_GAIN_MAX, DA7219_NO_INVERT,
+ da7219_volsw_locked_get, da7219_volsw_locked_put,
+ da7219_hp_gain_tlv),
+ SOC_DOUBLE_R_EXT("Headphone Switch", DA7219_HP_L_CTRL, DA7219_HP_R_CTRL,
+ DA7219_HP_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+ SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7219_HP_L_CTRL,
+ DA7219_HP_R_CTRL, DA7219_HP_L_AMP_RAMP_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+ SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7219_HP_L_CTRL,
+ DA7219_HP_R_CTRL, DA7219_HP_L_AMP_ZC_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+
+/*
+ * DAPM Mux Controls
+ */
+
+static const char * const da7219_out_sel_txt[] = {
+ "ADC", "Tone Generator", "DAIL", "DAIR"
+};
+
+static const struct soc_enum da7219_out_dail_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI,
+ DA7219_DAI_L_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dail_sel_mux =
+ SOC_DAPM_ENUM("Out DAIL Mux", da7219_out_dail_sel);
+
+static const struct soc_enum da7219_out_dair_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI,
+ DA7219_DAI_R_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dair_sel_mux =
+ SOC_DAPM_ENUM("Out DAIR Mux", da7219_out_dair_sel);
+
+static const struct soc_enum da7219_out_dacl_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC,
+ DA7219_DAC_L_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dacl_sel_mux =
+ SOC_DAPM_ENUM("Out DACL Mux", da7219_out_dacl_sel);
+
+static const struct soc_enum da7219_out_dacr_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC,
+ DA7219_DAC_R_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dacr_sel_mux =
+ SOC_DAPM_ENUM("Out DACR Mux", da7219_out_dacr_sel);
+
+
+/*
+ * DAPM Mixer Controls
+ */
+
+static const struct snd_kcontrol_new da7219_mixin_controls[] = {
+ SOC_DAPM_SINGLE("Mic Switch", DA7219_MIXIN_L_SELECT,
+ DA7219_MIXIN_L_MIX_SELECT_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+static const struct snd_kcontrol_new da7219_mixout_l_controls[] = {
+ SOC_DAPM_SINGLE("DACL Switch", DA7219_MIXOUT_L_SELECT,
+ DA7219_MIXOUT_L_MIX_SELECT_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+static const struct snd_kcontrol_new da7219_mixout_r_controls[] = {
+ SOC_DAPM_SINGLE("DACR Switch", DA7219_MIXOUT_R_SELECT,
+ DA7219_MIXOUT_R_MIX_SELECT_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+#define DA7219_DMIX_ST_CTRLS(reg) \
+ SOC_DAPM_SINGLE("Out FilterL Switch", reg, \
+ DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT, \
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \
+ SOC_DAPM_SINGLE("Out FilterR Switch", reg, \
+ DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT, \
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \
+ SOC_DAPM_SINGLE("Sidetone Switch", reg, \
+ DA7219_DMIX_ST_SRC_SIDETONE_SHIFT, \
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT) \
+
+static const struct snd_kcontrol_new da7219_st_out_filtl_mix_controls[] = {
+ DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1L),
+};
+
+static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = {
+ DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1R),
+};
+
+
+/*
+ * DAPM Events
+ */
+
+static int da7219_dai_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 pll_ctrl, pll_status;
+ int i = 0;
+ bool srm_lock = false;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (da7219->master)
+ /* Enable DAI clks for master mode */
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_CLK_EN_MASK,
+ DA7219_DAI_CLK_EN_MASK);
+
+ /* PC synchronised to DAI */
+ snd_soc_update_bits(codec, DA7219_PC_COUNT,
+ DA7219_PC_FREERUN_MASK, 0);
+
+ /* Slave mode, if SRM not enabled no need for status checks */
+ pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL);
+ if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM)
+ return 0;
+
+ /* Check SRM has locked */
+ do {
+ pll_status = snd_soc_read(codec, DA7219_PLL_SRM_STS);
+ if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) {
+ srm_lock = true;
+ } else {
+ ++i;
+ msleep(50);
+ }
+ } while ((i < DA7219_SRM_CHECK_RETRIES) & (!srm_lock));
+
+ if (!srm_lock)
+ dev_warn(codec->dev, "SRM failed to lock\n");
+
+ return 0;
+ case SND_SOC_DAPM_POST_PMD:
+ /* PC free-running */
+ snd_soc_update_bits(codec, DA7219_PC_COUNT,
+ DA7219_PC_FREERUN_MASK,
+ DA7219_PC_FREERUN_MASK);
+
+ /* Disable DAI clks if in master mode */
+ if (da7219->master)
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_CLK_EN_MASK, 0);
+ return 0;
+ default:
+ return -EINVAL;
+ }
+}
+
+
+/*
+ * DAPM Widgets
+ */
+
+static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = {
+ /* Input Supplies */
+ SND_SOC_DAPM_SUPPLY("Mic Bias", DA7219_MICBIAS_CTRL,
+ DA7219_MICBIAS1_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MIC"),
+
+ /* Input PGAs */
+ SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL,
+ DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+
+ /* Input Filters */
+ SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT,
+ DA7219_NO_INVERT),
+
+ /* Tone Generator */
+ SND_SOC_DAPM_SIGGEN("TONE"),
+ SND_SOC_DAPM_PGA("Tone Generator", DA7219_TONE_GEN_CFG1,
+ DA7219_START_STOPN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+
+ /* Sidetone Input */
+ SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7219_SIDETONE_CTRL,
+ DA7219_SIDETONE_EN_SHIFT, DA7219_NO_INVERT),
+
+ /* Input Mixer Supply */
+ SND_SOC_DAPM_SUPPLY("Mixer In Supply", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_MIX_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+
+ /* Input Mixer */
+ SND_SOC_DAPM_MIXER("Mixer In", SND_SOC_NOPM, 0, 0,
+ da7219_mixin_controls,
+ ARRAY_SIZE(da7219_mixin_controls)),
+
+ /* Input Muxes */
+ SND_SOC_DAPM_MUX("Out DAIL Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dail_sel_mux),
+ SND_SOC_DAPM_MUX("Out DAIR Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dair_sel_mux),
+
+ /* DAI Supply */
+ SND_SOC_DAPM_SUPPLY("DAI", DA7219_DAI_CTRL, DA7219_DAI_EN_SHIFT,
+ DA7219_NO_INVERT, da7219_dai_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* DAI */
+ SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ /* Output Muxes */
+ SND_SOC_DAPM_MUX("Out DACL Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dacl_sel_mux),
+ SND_SOC_DAPM_MUX("Out DACR Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dacr_sel_mux),
+
+ /* Output Mixers */
+ SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0,
+ da7219_mixout_l_controls,
+ ARRAY_SIZE(da7219_mixout_l_controls)),
+ SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0,
+ da7219_mixout_r_controls,
+ ARRAY_SIZE(da7219_mixout_r_controls)),
+
+ /* Sidetone Mixers */
+ SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0,
+ da7219_st_out_filtl_mix_controls,
+ ARRAY_SIZE(da7219_st_out_filtl_mix_controls)),
+ SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0,
+ 0, da7219_st_out_filtr_mix_controls,
+ ARRAY_SIZE(da7219_st_out_filtr_mix_controls)),
+
+ /* DACs */
+ SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT,
+ DA7219_NO_INVERT),
+ SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT,
+ DA7219_NO_INVERT),
+
+ /* Output PGAs */
+ SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL,
+ DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL,
+ DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+ SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+
+ /* Output Supplies */
+ SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT,
+ DA7219_NO_INVERT, NULL, 0),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+
+/*
+ * DAPM Mux Routes
+ */
+
+#define DA7219_OUT_DAI_MUX_ROUTES(name) \
+ {name, "ADC", "Mixer In"}, \
+ {name, "Tone Generator", "Tone Generator"}, \
+ {name, "DAIL", "DAIOUT"}, \
+ {name, "DAIR", "DAIOUT"}
+
+#define DA7219_OUT_DAC_MUX_ROUTES(name) \
+ {name, "ADC", "Mixer In"}, \
+ {name, "Tone Generator", "Tone Generator"}, \
+ {name, "DAIL", "DAIIN"}, \
+ {name, "DAIR", "DAIIN"}
+
+/*
+ * DAPM Mixer Routes
+ */
+
+#define DA7219_DMIX_ST_ROUTES(name) \
+ {name, "Out FilterL Switch", "Mixer Out FilterL"}, \
+ {name, "Out FilterR Switch", "Mixer Out FilterR"}, \
+ {name, "Sidetone Switch", "Sidetone Filter"}
+
+
+/*
+ * DAPM audio route definition
+ */
+
+static const struct snd_soc_dapm_route da7219_audio_map[] = {
+ /* Input paths */
+ {"MIC", NULL, "Mic Bias"},
+ {"Mic PGA", NULL, "MIC"},
+ {"Mixin PGA", NULL, "Mic PGA"},
+ {"ADC", NULL, "Mixin PGA"},
+
+ {"Sidetone Filter", NULL, "ADC"},
+ {"Mixer In", NULL, "Mixer In Supply"},
+ {"Mixer In", "Mic Switch", "ADC"},
+
+ {"Tone Generator", NULL, "TONE"},
+
+ DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"),
+ DA7219_OUT_DAI_MUX_ROUTES("Out DAIR Mux"),
+
+ {"DAIOUT", NULL, "Out DAIL Mux"},
+ {"DAIOUT", NULL, "Out DAIR Mux"},
+ {"DAIOUT", NULL, "DAI"},
+
+ /* Output paths */
+ {"DAIIN", NULL, "DAI"},
+
+ DA7219_OUT_DAC_MUX_ROUTES("Out DACL Mux"),
+ DA7219_OUT_DAC_MUX_ROUTES("Out DACR Mux"),
+
+ {"Mixer Out FilterL", "DACL Switch", "Out DACL Mux"},
+ {"Mixer Out FilterR", "DACR Switch", "Out DACR Mux"},
+
+ DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterL"),
+ DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterR"),
+
+ {"DACL", NULL, "ST Mixer Out FilterL"},
+ {"DACR", NULL, "ST Mixer Out FilterR"},
+
+ {"Mixout Left PGA", NULL, "DACL"},
+ {"Mixout Right PGA", NULL, "DACR"},
+
+ {"Headphone Left PGA", NULL, "Mixout Left PGA"},
+ {"Headphone Right PGA", NULL, "Mixout Right PGA"},
+
+ {"HPL", NULL, "Headphone Left PGA"},
+ {"HPR", NULL, "Headphone Right PGA"},
+
+ {"HPL", NULL, "Charge Pump"},
+ {"HPR", NULL, "Charge Pump"},
+};
+
+
+/*
+ * DAI operations
+ */
+
+static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq))
+ return 0;
+
+ if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) {
+ dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+ freq);
+ return -EINVAL;
+ }
+
+ switch (clk_id) {
+ case DA7219_CLKSRC_MCLK_SQR:
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_MCLK_SQR_EN_MASK,
+ DA7219_PLL_MCLK_SQR_EN_MASK);
+ break;
+ case DA7219_CLKSRC_MCLK:
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_MCLK_SQR_EN_MASK, 0);
+ break;
+ default:
+ dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
+ return -EINVAL;
+ }
+
+ da7219->clk_src = clk_id;
+
+ if (da7219->mclk) {
+ freq = clk_round_rate(da7219->mclk, freq);
+ ret = clk_set_rate(da7219->mclk, freq);
+ if (ret) {
+ dev_err(codec_dai->dev, "Failed to set clock rate %d\n",
+ freq);
+ return ret;
+ }
+ }
+
+ da7219->mclk_rate = freq;
+
+ return 0;
+}
+
+static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int fref, unsigned int fout)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ u8 pll_ctrl, indiv_bits, indiv;
+ u8 pll_frac_top, pll_frac_bot, pll_integer;
+ u32 freq_ref;
+ u64 frac_div;
+
+ /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */
+ if (da7219->mclk_rate == 32768) {
+ indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
+ indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
+ } else if (da7219->mclk_rate < 2000000) {
+ dev_err(codec->dev, "PLL input clock %d below valid range\n",
+ da7219->mclk_rate);
+ return -EINVAL;
+ } else if (da7219->mclk_rate <= 5000000) {
+ indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
+ indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 10000000) {
+ indiv_bits = DA7219_PLL_INDIV_5_10_MHZ;
+ indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 20000000) {
+ indiv_bits = DA7219_PLL_INDIV_10_20_MHZ;
+ indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 40000000) {
+ indiv_bits = DA7219_PLL_INDIV_20_40_MHZ;
+ indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 54000000) {
+ indiv_bits = DA7219_PLL_INDIV_40_54_MHZ;
+ indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL;
+ } else {
+ dev_err(codec->dev, "PLL input clock %d above valid range\n",
+ da7219->mclk_rate);
+ return -EINVAL;
+ }
+ freq_ref = (da7219->mclk_rate / indiv);
+ pll_ctrl = indiv_bits;
+
+ /* Configure PLL */
+ switch (source) {
+ case DA7219_SYSCLK_MCLK:
+ pll_ctrl |= DA7219_PLL_MODE_BYPASS;
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_INDIV_MASK |
+ DA7219_PLL_MODE_MASK, pll_ctrl);
+ return 0;
+ case DA7219_SYSCLK_PLL:
+ pll_ctrl |= DA7219_PLL_MODE_NORMAL;
+ break;
+ case DA7219_SYSCLK_PLL_SRM:
+ pll_ctrl |= DA7219_PLL_MODE_SRM;
+ break;
+ case DA7219_SYSCLK_PLL_32KHZ:
+ pll_ctrl |= DA7219_PLL_MODE_32KHZ;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid PLL config\n");
+ return -EINVAL;
+ }
+
+ /* Calculate dividers for PLL */
+ pll_integer = fout / freq_ref;
+ frac_div = (u64)(fout % freq_ref) * 8192ULL;
+ do_div(frac_div, freq_ref);
+ pll_frac_top = (frac_div >> DA7219_BYTE_SHIFT) & DA7219_BYTE_MASK;
+ pll_frac_bot = (frac_div) & DA7219_BYTE_MASK;
+
+ /* Write PLL config & dividers */
+ snd_soc_write(codec, DA7219_PLL_FRAC_TOP, pll_frac_top);
+ snd_soc_write(codec, DA7219_PLL_FRAC_BOT, pll_frac_bot);
+ snd_soc_write(codec, DA7219_PLL_INTEGER, pll_integer);
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK,
+ pll_ctrl);
+
+ return 0;
+}
+
+static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 dai_clk_mode = 0, dai_ctrl = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ da7219->master = true;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ da7219->master = false;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ dai_clk_mode |= DA7219_DAI_WCLK_POL_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ dai_clk_mode |= DA7219_DAI_CLK_POL_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ dai_clk_mode |= DA7219_DAI_WCLK_POL_INV |
+ DA7219_DAI_CLK_POL_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ dai_ctrl |= DA7219_DAI_FORMAT_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ dai_ctrl |= DA7219_DAI_FORMAT_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ dai_ctrl |= DA7219_DAI_FORMAT_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ dai_ctrl |= DA7219_DAI_FORMAT_DSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* By default 64 BCLKs per WCLK is supported */
+ dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64;
+
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_BCLKS_PER_WCLK_MASK |
+ DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK,
+ dai_clk_mode);
+ snd_soc_update_bits(codec, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK,
+ dai_ctrl);
+
+ return 0;
+}
+
+static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 dai_bclks_per_wclk;
+ u16 offset;
+ u32 frame_size;
+
+ /* No channels enabled so disable TDM, revert to 64-bit frames */
+ if (!tx_mask) {
+ snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL,
+ DA7219_DAI_TDM_CH_EN_MASK |
+ DA7219_DAI_TDM_MODE_EN_MASK, 0);
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_BCLKS_PER_WCLK_MASK,
+ DA7219_DAI_BCLKS_PER_WCLK_64);
+ return 0;
+ }
+
+ /* Check we have valid slots */
+ if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) {
+ dev_err(codec->dev, "Invalid number of slots, max = %d\n",
+ DA7219_DAI_TDM_MAX_SLOTS);
+ return -EINVAL;
+ }
+
+ /* Check we have a valid offset given */
+ if (rx_mask > DA7219_DAI_OFFSET_MAX) {
+ dev_err(codec->dev, "Invalid slot offset, max = %d\n",
+ DA7219_DAI_OFFSET_MAX);
+ return -EINVAL;
+ }
+
+ /* Calculate & validate frame size based on slot info provided. */
+ frame_size = slots * slot_width;
+ switch (frame_size) {
+ case 32:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32;
+ break;
+ case 64:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64;
+ break;
+ case 128:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128;
+ break;
+ case 256:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid frame size %d\n", frame_size);
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_BCLKS_PER_WCLK_MASK,
+ dai_bclks_per_wclk);
+
+ offset = cpu_to_le16(rx_mask);
+ regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER,
+ &offset, sizeof(offset));
+
+ snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL,
+ DA7219_DAI_TDM_CH_EN_MASK |
+ DA7219_DAI_TDM_MODE_EN_MASK,
+ (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) |
+ DA7219_DAI_TDM_MODE_EN_MASK);
+
+ return 0;
+}
+
+static int da7219_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 dai_ctrl = 0, fs;
+ unsigned int channels;
+
+ switch (params_width(params)) {
+ case 16:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE;
+ break;
+ case 20:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE;
+ break;
+ case 24:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE;
+ break;
+ case 32:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ channels = params_channels(params);
+ if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) {
+ dev_err(codec->dev,
+ "Invalid number of channels, only 1 to %d supported\n",
+ DA7219_DAI_CH_NUM_MAX);
+ return -EINVAL;
+ }
+ dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT;
+
+ switch (params_rate(params)) {
+ case 8000:
+ fs = DA7219_SR_8000;
+ break;
+ case 11025:
+ fs = DA7219_SR_11025;
+ break;
+ case 12000:
+ fs = DA7219_SR_12000;
+ break;
+ case 16000:
+ fs = DA7219_SR_16000;
+ break;
+ case 22050:
+ fs = DA7219_SR_22050;
+ break;
+ case 24000:
+ fs = DA7219_SR_24000;
+ break;
+ case 32000:
+ fs = DA7219_SR_32000;
+ break;
+ case 44100:
+ fs = DA7219_SR_44100;
+ break;
+ case 48000:
+ fs = DA7219_SR_48000;
+ break;
+ case 88200:
+ fs = DA7219_SR_88200;
+ break;
+ case 96000:
+ fs = DA7219_SR_96000;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, DA7219_DAI_CTRL,
+ DA7219_DAI_WORD_LENGTH_MASK |
+ DA7219_DAI_CH_NUM_MASK,
+ dai_ctrl);
+ snd_soc_write(codec, DA7219_SR, fs);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops da7219_dai_ops = {
+ .hw_params = da7219_hw_params,
+ .set_sysclk = da7219_set_dai_sysclk,
+ .set_pll = da7219_set_dai_pll,
+ .set_fmt = da7219_set_dai_fmt,
+ .set_tdm_slot = da7219_set_dai_tdm_slot,
+};
+
+#define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver da7219_dai = {
+ .name = "da7219-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = DA7219_DAI_CH_NUM_MAX,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = DA7219_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = DA7219_DAI_CH_NUM_MAX,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = DA7219_FORMATS,
+ },
+ .ops = &da7219_dai_ops,
+ .symmetric_rates = 1,
+ .symmetric_channels = 1,
+ .symmetric_samplebits = 1,
+};
+
+
+/*
+ * DT
+ */
+
+static const struct of_device_id da7219_of_match[] = {
+ { .compatible = "dlg,da7219", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, da7219_of_match);
+
+static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec,
+ u32 val)
+{
+ switch (val) {
+ case 1050:
+ return DA7219_LDO_LVL_SEL_1_05V;
+ case 1100:
+ return DA7219_LDO_LVL_SEL_1_10V;
+ case 1200:
+ return DA7219_LDO_LVL_SEL_1_20V;
+ case 1400:
+ return DA7219_LDO_LVL_SEL_1_40V;
+ default:
+ dev_warn(codec->dev, "Invalid LDO level");
+ return DA7219_LDO_LVL_SEL_1_05V;
+ }
+}
+
+static enum da7219_micbias_voltage
+ da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1800:
+ return DA7219_MICBIAS_1_8V;
+ case 2000:
+ return DA7219_MICBIAS_2_0V;
+ case 2200:
+ return DA7219_MICBIAS_2_2V;
+ case 2400:
+ return DA7219_MICBIAS_2_4V;
+ case 2600:
+ return DA7219_MICBIAS_2_6V;
+ default:
+ dev_warn(codec->dev, "Invalid micbias level");
+ return DA7219_MICBIAS_2_2V;
+ }
+}
+
+static enum da7219_mic_amp_in_sel
+ da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "diff")) {
+ return DA7219_MIC_AMP_IN_SEL_DIFF;
+ } else if (!strcmp(str, "se_p")) {
+ return DA7219_MIC_AMP_IN_SEL_SE_P;
+ } else if (!strcmp(str, "se_n")) {
+ return DA7219_MIC_AMP_IN_SEL_SE_N;
+ } else {
+ dev_warn(codec->dev, "Invalid mic input type selection");
+ return DA7219_MIC_AMP_IN_SEL_DIFF;
+ }
+}
+
+static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec)
+{
+ struct device_node *np = codec->dev->of_node;
+ struct da7219_pdata *pdata;
+ const char *of_str;
+ u32 of_val32;
+
+ pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata)
+ return NULL;
+
+ if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0)
+ pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32);
+
+ if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0)
+ pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32);
+ else
+ pdata->micbias_lvl = DA7219_MICBIAS_2_2V;
+
+ if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str))
+ pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str);
+ else
+ pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF;
+
+ return pdata;
+}
+
+
+/*
+ * Codec driver functions
+ */
+
+static int da7219_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+ /* MCLK */
+ if (da7219->mclk) {
+ ret = clk_prepare_enable(da7219->mclk);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to enable mclk\n");
+ return ret;
+ }
+ }
+
+ /* Master bias */
+ snd_soc_update_bits(codec, DA7219_REFERENCES,
+ DA7219_BIAS_EN_MASK,
+ DA7219_BIAS_EN_MASK);
+
+ /* Enable Internal Digital LDO */
+ snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+ DA7219_LDO_EN_MASK,
+ DA7219_LDO_EN_MASK);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* Only disable if jack detection not active */
+ if (!da7219->aad->jack) {
+ /* Bypass Internal Digital LDO */
+ snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+ DA7219_LDO_EN_MASK, 0);
+
+ /* Master bias */
+ snd_soc_update_bits(codec, DA7219_REFERENCES,
+ DA7219_BIAS_EN_MASK, 0);
+ }
+
+ /* MCLK */
+ if (da7219->mclk)
+ clk_disable_unprepare(da7219->mclk);
+ break;
+ }
+
+ return 0;
+}
+
+static const char *da7219_supply_names[DA7219_NUM_SUPPLIES] = {
+ [DA7219_SUPPLY_VDD] = "VDD",
+ [DA7219_SUPPLY_VDDMIC] = "VDDMIC",
+ [DA7219_SUPPLY_VDDIO] = "VDDIO",
+};
+
+static int da7219_handle_supplies(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct regulator *vddio;
+ u8 io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V;
+ int i, ret;
+
+ /* Get required supplies */
+ for (i = 0; i < DA7219_NUM_SUPPLIES; ++i)
+ da7219->supplies[i].supply = da7219_supply_names[i];
+
+ ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES,
+ da7219->supplies);
+ if (ret) {
+ dev_err(codec->dev, "Failed to get supplies");
+ return ret;
+ }
+
+ /* Determine VDDIO voltage provided */
+ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer;
+ ret = regulator_get_voltage(vddio);
+ if (ret < 1200000)
+ dev_warn(codec->dev, "Invalid VDDIO voltage\n");
+ else if (ret < 2800000)
+ io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V;
+
+ /* Enable main supplies */
+ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies);
+ if (ret) {
+ dev_err(codec->dev, "Failed to enable supplies");
+ return ret;
+ }
+
+ /* Ensure device in active mode */
+ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK);
+
+ /* Update IO voltage level range */
+ snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl);
+
+ return 0;
+}
+
+static void da7219_handle_pdata(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_pdata *pdata = da7219->pdata;
+
+ if (pdata) {
+ u8 micbias_lvl = 0;
+
+ /* Internal LDO */
+ switch (pdata->ldo_lvl_sel) {
+ case DA7219_LDO_LVL_SEL_1_05V:
+ case DA7219_LDO_LVL_SEL_1_10V:
+ case DA7219_LDO_LVL_SEL_1_20V:
+ case DA7219_LDO_LVL_SEL_1_40V:
+ snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+ DA7219_LDO_LEVEL_SELECT_MASK,
+ (pdata->ldo_lvl_sel <<
+ DA7219_LDO_LEVEL_SELECT_SHIFT));
+ break;
+ }
+
+ /* Mic Bias voltages */
+ switch (pdata->micbias_lvl) {
+ case DA7219_MICBIAS_1_8V:
+ case DA7219_MICBIAS_2_0V:
+ case DA7219_MICBIAS_2_2V:
+ case DA7219_MICBIAS_2_4V:
+ case DA7219_MICBIAS_2_6V:
+ micbias_lvl |= (pdata->micbias_lvl <<
+ DA7219_MICBIAS1_LEVEL_SHIFT);
+ break;
+ }
+
+ snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_lvl);
+
+ /* Mic */
+ switch (pdata->mic_amp_in_sel) {
+ case DA7219_MIC_AMP_IN_SEL_DIFF:
+ case DA7219_MIC_AMP_IN_SEL_SE_P:
+ case DA7219_MIC_AMP_IN_SEL_SE_N:
+ snd_soc_write(codec, DA7219_MIC_1_SELECT,
+ pdata->mic_amp_in_sel);
+ break;
+ }
+ }
+}
+
+static int da7219_probe(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_init(&da7219->lock);
+
+ /* Regulator configuration */
+ ret = da7219_handle_supplies(codec);
+ if (ret)
+ return ret;
+
+ /* Handle DT/Platform data */
+ if (codec->dev->of_node)
+ da7219->pdata = da7219_of_to_pdata(codec);
+ else
+ da7219->pdata = dev_get_platdata(codec->dev);
+
+ da7219_handle_pdata(codec);
+
+ /* Check if MCLK provided */
+ da7219->mclk = devm_clk_get(codec->dev, "mclk");
+ if (IS_ERR(da7219->mclk)) {
+ if (PTR_ERR(da7219->mclk) != -ENOENT)
+ return PTR_ERR(da7219->mclk);
+ else
+ da7219->mclk = NULL;
+ }
+
+ /* Default PC counter to free-running */
+ snd_soc_update_bits(codec, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK,
+ DA7219_PC_FREERUN_MASK);
+
+ /* Default gain ramping */
+ snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_RAMP_EN_MASK,
+ DA7219_MIXIN_L_AMP_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK,
+ DA7219_ADC_L_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK,
+ DA7219_DAC_L_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK,
+ DA7219_DAC_R_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_RAMP_EN_MASK,
+ DA7219_HP_L_AMP_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_RAMP_EN_MASK,
+ DA7219_HP_R_AMP_RAMP_EN_MASK);
+
+ /* Default infinite tone gen, start/stop by Kcontrol */
+ snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK);
+
+ /* Initialise AAD block */
+ return da7219_aad_init(codec);
+}
+
+static int da7219_remove(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ da7219_aad_exit(codec);
+
+ /* Supplies */
+ return regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies);
+}
+
+#ifdef CONFIG_PM
+static int da7219_suspend(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ /* Put device into standby mode if jack detection disabled */
+ if (!da7219->aad->jack)
+ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0);
+
+ return 0;
+}
+
+static int da7219_resume(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ /* Put device into active mode if previously pushed to standby */
+ if (!da7219->aad->jack)
+ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE,
+ DA7219_SYSTEM_ACTIVE_MASK);
+
+ snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define da7219_suspend NULL
+#define da7219_resume NULL
+#endif
+
+static struct snd_soc_codec_driver soc_codec_dev_da7219 = {
+ .probe = da7219_probe,
+ .remove = da7219_remove,
+ .suspend = da7219_suspend,
+ .resume = da7219_resume,
+ .set_bias_level = da7219_set_bias_level,
+
+ .controls = da7219_snd_controls,
+ .num_controls = ARRAY_SIZE(da7219_snd_controls),
+
+ .dapm_widgets = da7219_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(da7219_dapm_widgets),
+ .dapm_routes = da7219_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(da7219_audio_map),
+};
+
+
+/*
+ * Regmap configs
+ */
+
+static struct reg_default da7219_reg_defaults[] = {
+ { DA7219_MIC_1_SELECT, 0x00 },
+ { DA7219_CIF_TIMEOUT_CTRL, 0x01 },
+ { DA7219_SR_24_48, 0x00 },
+ { DA7219_SR, 0x0A },
+ { DA7219_CIF_I2C_ADDR_CFG, 0x02 },
+ { DA7219_PLL_CTRL, 0x10 },
+ { DA7219_PLL_FRAC_TOP, 0x00 },
+ { DA7219_PLL_FRAC_BOT, 0x00 },
+ { DA7219_PLL_INTEGER, 0x20 },
+ { DA7219_DIG_ROUTING_DAI, 0x10 },
+ { DA7219_DAI_CLK_MODE, 0x01 },
+ { DA7219_DAI_CTRL, 0x28 },
+ { DA7219_DAI_TDM_CTRL, 0x40 },
+ { DA7219_DIG_ROUTING_DAC, 0x32 },
+ { DA7219_DAI_OFFSET_LOWER, 0x00 },
+ { DA7219_DAI_OFFSET_UPPER, 0x00 },
+ { DA7219_REFERENCES, 0x00 },
+ { DA7219_MIXIN_L_SELECT, 0x00 },
+ { DA7219_MIXIN_L_GAIN, 0x03 },
+ { DA7219_ADC_L_GAIN, 0x6F },
+ { DA7219_ADC_FILTERS1, 0x80 },
+ { DA7219_MIC_1_GAIN, 0x01 },
+ { DA7219_SIDETONE_CTRL, 0x40 },
+ { DA7219_SIDETONE_GAIN, 0x0E },
+ { DA7219_DROUTING_ST_OUTFILT_1L, 0x01 },
+ { DA7219_DROUTING_ST_OUTFILT_1R, 0x02 },
+ { DA7219_DAC_FILTERS5, 0x00 },
+ { DA7219_DAC_FILTERS2, 0x88 },
+ { DA7219_DAC_FILTERS3, 0x88 },
+ { DA7219_DAC_FILTERS4, 0x08 },
+ { DA7219_DAC_FILTERS1, 0x80 },
+ { DA7219_DAC_L_GAIN, 0x6F },
+ { DA7219_DAC_R_GAIN, 0x6F },
+ { DA7219_CP_CTRL, 0x20 },
+ { DA7219_HP_L_GAIN, 0x39 },
+ { DA7219_HP_R_GAIN, 0x39 },
+ { DA7219_MIXOUT_L_SELECT, 0x00 },
+ { DA7219_MIXOUT_R_SELECT, 0x00 },
+ { DA7219_MICBIAS_CTRL, 0x03 },
+ { DA7219_MIC_1_CTRL, 0x40 },
+ { DA7219_MIXIN_L_CTRL, 0x40 },
+ { DA7219_ADC_L_CTRL, 0x40 },
+ { DA7219_DAC_L_CTRL, 0x40 },
+ { DA7219_DAC_R_CTRL, 0x40 },
+ { DA7219_HP_L_CTRL, 0x40 },
+ { DA7219_HP_R_CTRL, 0x40 },
+ { DA7219_MIXOUT_L_CTRL, 0x10 },
+ { DA7219_MIXOUT_R_CTRL, 0x10 },
+ { DA7219_CHIP_ID1, 0x23 },
+ { DA7219_CHIP_ID2, 0x93 },
+ { DA7219_CHIP_REVISION, 0x00 },
+ { DA7219_LDO_CTRL, 0x00 },
+ { DA7219_IO_CTRL, 0x00 },
+ { DA7219_GAIN_RAMP_CTRL, 0x00 },
+ { DA7219_PC_COUNT, 0x02 },
+ { DA7219_CP_VOL_THRESHOLD1, 0x0E },
+ { DA7219_DIG_CTRL, 0x00 },
+ { DA7219_ALC_CTRL2, 0x00 },
+ { DA7219_ALC_CTRL3, 0x00 },
+ { DA7219_ALC_NOISE, 0x3F },
+ { DA7219_ALC_TARGET_MIN, 0x3F },
+ { DA7219_ALC_TARGET_MAX, 0x00 },
+ { DA7219_ALC_GAIN_LIMITS, 0xFF },
+ { DA7219_ALC_ANA_GAIN_LIMITS, 0x71 },
+ { DA7219_ALC_ANTICLIP_CTRL, 0x00 },
+ { DA7219_ALC_ANTICLIP_LEVEL, 0x00 },
+ { DA7219_DAC_NG_SETUP_TIME, 0x00 },
+ { DA7219_DAC_NG_OFF_THRESH, 0x00 },
+ { DA7219_DAC_NG_ON_THRESH, 0x00 },
+ { DA7219_DAC_NG_CTRL, 0x00 },
+ { DA7219_TONE_GEN_CFG1, 0x00 },
+ { DA7219_TONE_GEN_CFG2, 0x00 },
+ { DA7219_TONE_GEN_CYCLES, 0x00 },
+ { DA7219_TONE_GEN_FREQ1_L, 0x55 },
+ { DA7219_TONE_GEN_FREQ1_U, 0x15 },
+ { DA7219_TONE_GEN_FREQ2_L, 0x00 },
+ { DA7219_TONE_GEN_FREQ2_U, 0x40 },
+ { DA7219_TONE_GEN_ON_PER, 0x02 },
+ { DA7219_TONE_GEN_OFF_PER, 0x01 },
+ { DA7219_ACCDET_IRQ_MASK_A, 0x00 },
+ { DA7219_ACCDET_IRQ_MASK_B, 0x00 },
+ { DA7219_ACCDET_CONFIG_1, 0xD6 },
+ { DA7219_ACCDET_CONFIG_2, 0x34 },
+ { DA7219_ACCDET_CONFIG_3, 0x0A },
+ { DA7219_ACCDET_CONFIG_4, 0x16 },
+ { DA7219_ACCDET_CONFIG_5, 0x21 },
+ { DA7219_ACCDET_CONFIG_6, 0x3E },
+ { DA7219_ACCDET_CONFIG_7, 0x01 },
+ { DA7219_SYSTEM_ACTIVE, 0x00 },
+};
+
+static bool da7219_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case DA7219_MIC_1_GAIN_STATUS:
+ case DA7219_MIXIN_L_GAIN_STATUS:
+ case DA7219_ADC_L_GAIN_STATUS:
+ case DA7219_DAC_L_GAIN_STATUS:
+ case DA7219_DAC_R_GAIN_STATUS:
+ case DA7219_HP_L_GAIN_STATUS:
+ case DA7219_HP_R_GAIN_STATUS:
+ case DA7219_CIF_CTRL:
+ case DA7219_PLL_SRM_STS:
+ case DA7219_ALC_CTRL1:
+ case DA7219_SYSTEM_MODES_INPUT:
+ case DA7219_SYSTEM_MODES_OUTPUT:
+ case DA7219_ALC_OFFSET_AUTO_M_L:
+ case DA7219_ALC_OFFSET_AUTO_U_L:
+ case DA7219_TONE_GEN_CFG1:
+ case DA7219_ACCDET_STATUS_A:
+ case DA7219_ACCDET_STATUS_B:
+ case DA7219_ACCDET_IRQ_EVENT_A:
+ case DA7219_ACCDET_IRQ_EVENT_B:
+ case DA7219_ACCDET_CONFIG_8:
+ case DA7219_SYSTEM_STATUS:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static const struct regmap_config da7219_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = DA7219_SYSTEM_ACTIVE,
+ .reg_defaults = da7219_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(da7219_reg_defaults),
+ .volatile_reg = da7219_volatile_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+
+/*
+ * I2C layer
+ */
+
+static int da7219_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct da7219_priv *da7219;
+ int ret;
+
+ da7219 = devm_kzalloc(&i2c->dev, sizeof(struct da7219_priv),
+ GFP_KERNEL);
+ if (!da7219)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, da7219);
+
+ da7219->regmap = devm_regmap_init_i2c(i2c, &da7219_regmap_config);
+ if (IS_ERR(da7219->regmap)) {
+ ret = PTR_ERR(da7219->regmap);
+ dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219,
+ &da7219_dai, 1);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register da7219 codec: %d\n",
+ ret);
+ }
+ return ret;
+}
+
+static int da7219_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id da7219_i2c_id[] = {
+ { "da7219", },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, da7219_i2c_id);
+
+static struct i2c_driver da7219_i2c_driver = {
+ .driver = {
+ .name = "da7219",
+ .of_match_table = of_match_ptr(da7219_of_match),
+ },
+ .probe = da7219_i2c_probe,
+ .remove = da7219_i2c_remove,
+ .id_table = da7219_i2c_id,
+};
+
+module_i2c_driver(da7219_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC DA7219 Codec Driver");
+MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
new file mode 100644
index 0000000..b514268
--- /dev/null
+++ b/sound/soc/codecs/da7219.h
@@ -0,0 +1,820 @@
+/*
+ * da7219.h - DA7219 ALSA SoC Codec Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_H
+#define __DA7219_H
+
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <sound/da7219.h>
+
+/*
+ * Registers
+ */
+
+#define DA7219_MIC_1_GAIN_STATUS 0x6
+#define DA7219_MIXIN_L_GAIN_STATUS 0x8
+#define DA7219_ADC_L_GAIN_STATUS 0xA
+#define DA7219_DAC_L_GAIN_STATUS 0xC
+#define DA7219_DAC_R_GAIN_STATUS 0xD
+#define DA7219_HP_L_GAIN_STATUS 0xE
+#define DA7219_HP_R_GAIN_STATUS 0xF
+#define DA7219_MIC_1_SELECT 0x10
+#define DA7219_CIF_TIMEOUT_CTRL 0x12
+#define DA7219_CIF_CTRL 0x13
+#define DA7219_SR_24_48 0x16
+#define DA7219_SR 0x17
+#define DA7219_CIF_I2C_ADDR_CFG 0x1B
+#define DA7219_PLL_CTRL 0x20
+#define DA7219_PLL_FRAC_TOP 0x22
+#define DA7219_PLL_FRAC_BOT 0x23
+#define DA7219_PLL_INTEGER 0x24
+#define DA7219_PLL_SRM_STS 0x25
+#define DA7219_DIG_ROUTING_DAI 0x2A
+#define DA7219_DAI_CLK_MODE 0x2B
+#define DA7219_DAI_CTRL 0x2C
+#define DA7219_DAI_TDM_CTRL 0x2D
+#define DA7219_DIG_ROUTING_DAC 0x2E
+#define DA7219_ALC_CTRL1 0x2F
+#define DA7219_DAI_OFFSET_LOWER 0x30
+#define DA7219_DAI_OFFSET_UPPER 0x31
+#define DA7219_REFERENCES 0x32
+#define DA7219_MIXIN_L_SELECT 0x33
+#define DA7219_MIXIN_L_GAIN 0x34
+#define DA7219_ADC_L_GAIN 0x36
+#define DA7219_ADC_FILTERS1 0x38
+#define DA7219_MIC_1_GAIN 0x39
+#define DA7219_SIDETONE_CTRL 0x3A
+#define DA7219_SIDETONE_GAIN 0x3B
+#define DA7219_DROUTING_ST_OUTFILT_1L 0x3C
+#define DA7219_DROUTING_ST_OUTFILT_1R 0x3D
+#define DA7219_DAC_FILTERS5 0x40
+#define DA7219_DAC_FILTERS2 0x41
+#define DA7219_DAC_FILTERS3 0x42
+#define DA7219_DAC_FILTERS4 0x43
+#define DA7219_DAC_FILTERS1 0x44
+#define DA7219_DAC_L_GAIN 0x45
+#define DA7219_DAC_R_GAIN 0x46
+#define DA7219_CP_CTRL 0x47
+#define DA7219_HP_L_GAIN 0x48
+#define DA7219_HP_R_GAIN 0x49
+#define DA7219_MIXOUT_L_SELECT 0x4B
+#define DA7219_MIXOUT_R_SELECT 0x4C
+#define DA7219_SYSTEM_MODES_INPUT 0x50
+#define DA7219_SYSTEM_MODES_OUTPUT 0x51
+#define DA7219_MICBIAS_CTRL 0x62
+#define DA7219_MIC_1_CTRL 0x63
+#define DA7219_MIXIN_L_CTRL 0x65
+#define DA7219_ADC_L_CTRL 0x67
+#define DA7219_DAC_L_CTRL 0x69
+#define DA7219_DAC_R_CTRL 0x6A
+#define DA7219_HP_L_CTRL 0x6B
+#define DA7219_HP_R_CTRL 0x6C
+#define DA7219_MIXOUT_L_CTRL 0x6E
+#define DA7219_MIXOUT_R_CTRL 0x6F
+#define DA7219_CHIP_ID1 0x81
+#define DA7219_CHIP_ID2 0x82
+#define DA7219_CHIP_REVISION 0x83
+#define DA7219_LDO_CTRL 0x90
+#define DA7219_IO_CTRL 0x91
+#define DA7219_GAIN_RAMP_CTRL 0x92
+#define DA7219_PC_COUNT 0x94
+#define DA7219_CP_VOL_THRESHOLD1 0x95
+#define DA7219_CP_DELAY 0x96
+#define DA7219_DIG_CTRL 0x99
+#define DA7219_ALC_CTRL2 0x9A
+#define DA7219_ALC_CTRL3 0x9B
+#define DA7219_ALC_NOISE 0x9C
+#define DA7219_ALC_TARGET_MIN 0x9D
+#define DA7219_ALC_TARGET_MAX 0x9E
+#define DA7219_ALC_GAIN_LIMITS 0x9F
+#define DA7219_ALC_ANA_GAIN_LIMITS 0xA0
+#define DA7219_ALC_ANTICLIP_CTRL 0xA1
+#define DA7219_ALC_ANTICLIP_LEVEL 0xA2
+#define DA7219_ALC_OFFSET_AUTO_M_L 0xA3
+#define DA7219_ALC_OFFSET_AUTO_U_L 0xA4
+#define DA7219_DAC_NG_SETUP_TIME 0xAF
+#define DA7219_DAC_NG_OFF_THRESH 0xB0
+#define DA7219_DAC_NG_ON_THRESH 0xB1
+#define DA7219_DAC_NG_CTRL 0xB2
+#define DA7219_TONE_GEN_CFG1 0xB4
+#define DA7219_TONE_GEN_CFG2 0xB5
+#define DA7219_TONE_GEN_CYCLES 0xB6
+#define DA7219_TONE_GEN_FREQ1_L 0xB7
+#define DA7219_TONE_GEN_FREQ1_U 0xB8
+#define DA7219_TONE_GEN_FREQ2_L 0xB9
+#define DA7219_TONE_GEN_FREQ2_U 0xBA
+#define DA7219_TONE_GEN_ON_PER 0xBB
+#define DA7219_TONE_GEN_OFF_PER 0xBC
+#define DA7219_SYSTEM_STATUS 0xE0
+#define DA7219_SYSTEM_ACTIVE 0xFD
+
+
+/*
+ * Bit Fields
+ */
+
+#define DA7219_SWITCH_EN_MAX 0x1
+
+/* DA7219_MIC_1_GAIN_STATUS = 0x6 */
+#define DA7219_MIC_1_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_MIC_1_AMP_GAIN_STATUS_MASK (0x7 << 0)
+#define DA7219_MIC_1_AMP_GAIN_MAX 0x7
+
+/* DA7219_MIXIN_L_GAIN_STATUS = 0x8 */
+#define DA7219_MIXIN_L_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_MIXIN_L_AMP_GAIN_STATUS_MASK (0xF << 0)
+
+/* DA7219_ADC_L_GAIN_STATUS = 0xA */
+#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_SHIFT 0
+#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0)
+
+/* DA7219_DAC_L_GAIN_STATUS = 0xC */
+#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_SHIFT 0
+#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0)
+
+/* DA7219_DAC_R_GAIN_STATUS = 0xD */
+#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_SHIFT 0
+#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_MASK (0x7F << 0)
+
+/* DA7219_HP_L_GAIN_STATUS = 0xE */
+#define DA7219_HP_L_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_HP_L_AMP_GAIN_STATUS_MASK (0x3F << 0)
+
+/* DA7219_HP_R_GAIN_STATUS = 0xF */
+#define DA7219_HP_R_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_HP_R_AMP_GAIN_STATUS_MASK (0x3F << 0)
+
+/* DA7219_MIC_1_SELECT = 0x10 */
+#define DA7219_MIC_1_AMP_IN_SEL_SHIFT 0
+#define DA7219_MIC_1_AMP_IN_SEL_MASK (0x3 << 0)
+
+/* DA7219_CIF_TIMEOUT_CTRL = 0x12 */
+#define DA7219_I2C_TIMEOUT_EN_SHIFT 0
+#define DA7219_I2C_TIMEOUT_EN_MASK (0x1 << 0)
+
+/* DA7219_CIF_CTRL = 0x13 */
+#define DA7219_CIF_I2C_WRITE_MODE_SHIFT 0
+#define DA7219_CIF_I2C_WRITE_MODE_MASK (0x1 << 0)
+#define DA7219_CIF_REG_SOFT_RESET_SHIFT 7
+#define DA7219_CIF_REG_SOFT_RESET_MASK (0x1 << 7)
+
+/* DA7219_SR_24_48 = 0x16 */
+#define DA7219_SR_24_48_SHIFT 0
+#define DA7219_SR_24_48_MASK (0x1 << 0)
+
+/* DA7219_SR = 0x17 */
+#define DA7219_SR_SHIFT 0
+#define DA7219_SR_MASK (0xF << 0)
+#define DA7219_SR_8000 (0x01 << 0)
+#define DA7219_SR_11025 (0x02 << 0)
+#define DA7219_SR_12000 (0x03 << 0)
+#define DA7219_SR_16000 (0x05 << 0)
+#define DA7219_SR_22050 (0x06 << 0)
+#define DA7219_SR_24000 (0x07 << 0)
+#define DA7219_SR_32000 (0x09 << 0)
+#define DA7219_SR_44100 (0x0A << 0)
+#define DA7219_SR_48000 (0x0B << 0)
+#define DA7219_SR_88200 (0x0E << 0)
+#define DA7219_SR_96000 (0x0F << 0)
+
+/* DA7219_CIF_I2C_ADDR_CFG = 0x1B */
+#define DA7219_CIF_I2C_ADDR_CFG_SHIFT 0
+#define DA7219_CIF_I2C_ADDR_CFG_MASK (0x3 << 0)
+
+/* DA7219_PLL_CTRL = 0x20 */
+#define DA7219_PLL_INDIV_SHIFT 2
+#define DA7219_PLL_INDIV_MASK (0x7 << 2)
+#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2)
+#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2)
+#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2)
+#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2)
+#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2)
+#define DA7219_PLL_MCLK_SQR_EN_SHIFT 5
+#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5)
+#define DA7219_PLL_MODE_SHIFT 6
+#define DA7219_PLL_MODE_MASK (0x3 << 6)
+#define DA7219_PLL_MODE_BYPASS (0x0 << 6)
+#define DA7219_PLL_MODE_NORMAL (0x1 << 6)
+#define DA7219_PLL_MODE_SRM (0x2 << 6)
+#define DA7219_PLL_MODE_32KHZ (0x3 << 6)
+
+/* DA7219_PLL_FRAC_TOP = 0x22 */
+#define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT 0
+#define DA7219_PLL_FBDIV_FRAC_TOP_MASK (0x1F << 0)
+
+/* DA7219_PLL_FRAC_BOT = 0x23 */
+#define DA7219_PLL_FBDIV_FRAC_BOT_SHIFT 0
+#define DA7219_PLL_FBDIV_FRAC_BOT_MASK (0xFF << 0)
+
+/* DA7219_PLL_INTEGER = 0x24 */
+#define DA7219_PLL_FBDIV_INTEGER_SHIFT 0
+#define DA7219_PLL_FBDIV_INTEGER_MASK (0x7F << 0)
+
+/* DA7219_PLL_SRM_STS = 0x25 */
+#define DA7219_PLL_SRM_STATE_SHIFT 0
+#define DA7219_PLL_SRM_STATE_MASK (0xF << 0)
+#define DA7219_PLL_SRM_STATUS_SHIFT 4
+#define DA7219_PLL_SRM_STATUS_MASK (0xF << 4)
+#define DA7219_PLL_SRM_STS_SRM_LOCK (0x1 << 7)
+
+/* DA7219_DIG_ROUTING_DAI = 0x2A */
+#define DA7219_DAI_L_SRC_SHIFT 0
+#define DA7219_DAI_L_SRC_MASK (0x3 << 0)
+#define DA7219_DAI_R_SRC_SHIFT 4
+#define DA7219_DAI_R_SRC_MASK (0x3 << 4)
+#define DA7219_OUT_SRC_MAX 4
+
+/* DA7219_DAI_CLK_MODE = 0x2B */
+#define DA7219_DAI_BCLKS_PER_WCLK_SHIFT 0
+#define DA7219_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_32 (0x0 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_64 (0x1 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_128 (0x2 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_256 (0x3 << 0)
+#define DA7219_DAI_CLK_POL_SHIFT 2
+#define DA7219_DAI_CLK_POL_MASK (0x1 << 2)
+#define DA7219_DAI_CLK_POL_INV (0x1 << 2)
+#define DA7219_DAI_WCLK_POL_SHIFT 3
+#define DA7219_DAI_WCLK_POL_MASK (0x1 << 3)
+#define DA7219_DAI_WCLK_POL_INV (0x1 << 3)
+#define DA7219_DAI_WCLK_TRI_STATE_SHIFT 4
+#define DA7219_DAI_WCLK_TRI_STATE_MASK (0x1 << 4)
+#define DA7219_DAI_CLK_EN_SHIFT 7
+#define DA7219_DAI_CLK_EN_MASK (0x1 << 7)
+
+/* DA7219_DAI_CTRL = 0x2C */
+#define DA7219_DAI_FORMAT_SHIFT 0
+#define DA7219_DAI_FORMAT_MASK (0x3 << 0)
+#define DA7219_DAI_FORMAT_I2S (0x0 << 0)
+#define DA7219_DAI_FORMAT_LEFT_J (0x1 << 0)
+#define DA7219_DAI_FORMAT_RIGHT_J (0x2 << 0)
+#define DA7219_DAI_FORMAT_DSP (0x3 << 0)
+#define DA7219_DAI_WORD_LENGTH_SHIFT 2
+#define DA7219_DAI_WORD_LENGTH_MASK (0x3 << 2)
+#define DA7219_DAI_WORD_LENGTH_S16_LE (0x0 << 2)
+#define DA7219_DAI_WORD_LENGTH_S20_LE (0x1 << 2)
+#define DA7219_DAI_WORD_LENGTH_S24_LE (0x2 << 2)
+#define DA7219_DAI_WORD_LENGTH_S32_LE (0x3 << 2)
+#define DA7219_DAI_CH_NUM_SHIFT 4
+#define DA7219_DAI_CH_NUM_MASK (0x3 << 4)
+#define DA7219_DAI_CH_NUM_MAX 2
+#define DA7219_DAI_EN_SHIFT 7
+#define DA7219_DAI_EN_MASK (0x1 << 7)
+
+/* DA7219_DAI_TDM_CTRL = 0x2D */
+#define DA7219_DAI_TDM_CH_EN_SHIFT 0
+#define DA7219_DAI_TDM_CH_EN_MASK (0x3 << 0)
+#define DA7219_DAI_OE_SHIFT 6
+#define DA7219_DAI_OE_MASK (0x1 << 6)
+#define DA7219_DAI_TDM_MODE_EN_SHIFT 7
+#define DA7219_DAI_TDM_MODE_EN_MASK (0x1 << 7)
+#define DA7219_DAI_TDM_MAX_SLOTS 2
+
+/* DA7219_DIG_ROUTING_DAC = 0x2E */
+#define DA7219_DAC_L_SRC_SHIFT 0
+#define DA7219_DAC_L_SRC_MASK (0x3 << 0)
+#define DA7219_DAC_L_SRC_TONEGEN (0x1 << 0)
+#define DA7219_DAC_L_MONO_SHIFT 3
+#define DA7219_DAC_L_MONO_MASK (0x1 << 3)
+#define DA7219_DAC_R_SRC_SHIFT 4
+#define DA7219_DAC_R_SRC_MASK (0x3 << 4)
+#define DA7219_DAC_R_SRC_TONEGEN (0x1 << 4)
+#define DA7219_DAC_R_MONO_SHIFT 7
+#define DA7219_DAC_R_MONO_MASK (0x1 << 7)
+
+/* DA7219_ALC_CTRL1 = 0x2F */
+#define DA7219_ALC_OFFSET_EN_SHIFT 0
+#define DA7219_ALC_OFFSET_EN_MASK (0x1 << 0)
+#define DA7219_ALC_SYNC_MODE_SHIFT 1
+#define DA7219_ALC_SYNC_MODE_MASK (0x1 << 1)
+#define DA7219_ALC_EN_SHIFT 3
+#define DA7219_ALC_EN_MASK (0x1 << 3)
+#define DA7219_ALC_AUTO_CALIB_EN_SHIFT 4
+#define DA7219_ALC_AUTO_CALIB_EN_MASK (0x1 << 4)
+#define DA7219_ALC_CALIB_OVERFLOW_SHIFT 5
+#define DA7219_ALC_CALIB_OVERFLOW_MASK (0x1 << 5)
+
+/* DA7219_DAI_OFFSET_LOWER = 0x30 */
+#define DA7219_DAI_OFFSET_LOWER_SHIFT 0
+#define DA7219_DAI_OFFSET_LOWER_MASK (0xFF << 0)
+
+/* DA7219_DAI_OFFSET_UPPER = 0x31 */
+#define DA7219_DAI_OFFSET_UPPER_SHIFT 0
+#define DA7219_DAI_OFFSET_UPPER_MASK (0x7 << 0)
+#define DA7219_DAI_OFFSET_MAX 0x2FF
+
+/* DA7219_REFERENCES = 0x32 */
+#define DA7219_BIAS_EN_SHIFT 3
+#define DA7219_BIAS_EN_MASK (0x1 << 3)
+#define DA7219_VMID_FAST_CHARGE_SHIFT 4
+#define DA7219_VMID_FAST_CHARGE_MASK (0x1 << 4)
+
+/* DA7219_MIXIN_L_SELECT = 0x33 */
+#define DA7219_MIXIN_L_MIX_SELECT_SHIFT 0
+#define DA7219_MIXIN_L_MIX_SELECT_MASK (0x1 << 0)
+
+/* DA7219_MIXIN_L_GAIN = 0x34 */
+#define DA7219_MIXIN_L_AMP_GAIN_SHIFT 0
+#define DA7219_MIXIN_L_AMP_GAIN_MASK (0xF << 0)
+#define DA7219_MIXIN_L_AMP_GAIN_MAX 0xF
+
+/* DA7219_ADC_L_GAIN = 0x36 */
+#define DA7219_ADC_L_DIGITAL_GAIN_SHIFT 0
+#define DA7219_ADC_L_DIGITAL_GAIN_MASK (0x7F << 0)
+#define DA7219_ADC_L_DIGITAL_GAIN_MAX 0x7F
+
+/* DA7219_ADC_FILTERS1 = 0x38 */
+#define DA7219_ADC_VOICE_HPF_CORNER_SHIFT 0
+#define DA7219_ADC_VOICE_HPF_CORNER_MASK (0x7 << 0)
+#define DA7219_VOICE_HPF_CORNER_MAX 8
+#define DA7219_ADC_VOICE_EN_SHIFT 3
+#define DA7219_ADC_VOICE_EN_MASK (0x1 << 3)
+#define DA7219_ADC_AUDIO_HPF_CORNER_SHIFT 4
+#define DA7219_ADC_AUDIO_HPF_CORNER_MASK (0x3 << 4)
+#define DA7219_AUDIO_HPF_CORNER_MAX 4
+#define DA7219_ADC_HPF_EN_SHIFT 7
+#define DA7219_ADC_HPF_EN_MASK (0x1 << 7)
+#define DA7219_HPF_MODE_SHIFT 0
+#define DA7219_HPF_DISABLED ((0x0 << 3) | (0x0 << 7))
+#define DA7219_HPF_AUDIO_EN ((0x0 << 3) | (0x1 << 7))
+#define DA7219_HPF_VOICE_EN ((0x1 << 3) | (0x1 << 7))
+#define DA7219_HPF_MODE_MASK ((0x1 << 3) | (0x1 << 7))
+#define DA7219_HPF_MODE_MAX 3
+
+/* DA7219_MIC_1_GAIN = 0x39 */
+#define DA7219_MIC_1_AMP_GAIN_SHIFT 0
+#define DA7219_MIC_1_AMP_GAIN_MASK (0x7 << 0)
+
+/* DA7219_SIDETONE_CTRL = 0x3A */
+#define DA7219_SIDETONE_MUTE_EN_SHIFT 6
+#define DA7219_SIDETONE_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_SIDETONE_EN_SHIFT 7
+#define DA7219_SIDETONE_EN_MASK (0x1 << 7)
+
+/* DA7219_SIDETONE_GAIN = 0x3B */
+#define DA7219_SIDETONE_GAIN_SHIFT 0
+#define DA7219_SIDETONE_GAIN_MASK (0xF << 0)
+#define DA7219_SIDETONE_GAIN_MAX 0xE
+
+/* DA7219_DROUTING_ST_OUTFILT_1L = 0x3C */
+#define DA7219_OUTFILT_ST_1L_SRC_SHIFT 0
+#define DA7219_OUTFILT_ST_1L_SRC_MASK (0x7 << 0)
+#define DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT 0
+#define DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT 1
+#define DA7219_DMIX_ST_SRC_SIDETONE_SHIFT 2
+#define DA7219_DMIX_ST_SRC_OUTFILT1L (0x1 << 0)
+#define DA7219_DMIX_ST_SRC_OUTFILT1R (0x1 << 1)
+
+/* DA7219_DROUTING_ST_OUTFILT_1R = 0x3D */
+#define DA7219_OUTFILT_ST_1R_SRC_SHIFT 0
+#define DA7219_OUTFILT_ST_1R_SRC_MASK (0x7 << 0)
+
+/* DA7219_DAC_FILTERS5 = 0x40 */
+#define DA7219_DAC_SOFTMUTE_RATE_SHIFT 4
+#define DA7219_DAC_SOFTMUTE_RATE_MASK (0x7 << 4)
+#define DA7219_DAC_SOFTMUTE_RATE_MAX 7
+#define DA7219_DAC_SOFTMUTE_EN_SHIFT 7
+#define DA7219_DAC_SOFTMUTE_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_FILTERS2 = 0x41 */
+#define DA7219_DAC_EQ_BAND1_SHIFT 0
+#define DA7219_DAC_EQ_BAND1_MASK (0xF << 0)
+#define DA7219_DAC_EQ_BAND2_SHIFT 4
+#define DA7219_DAC_EQ_BAND2_MASK (0xF << 4)
+#define DA7219_DAC_EQ_BAND_MAX 0xF
+
+/* DA7219_DAC_FILTERS3 = 0x42 */
+#define DA7219_DAC_EQ_BAND3_SHIFT 0
+#define DA7219_DAC_EQ_BAND3_MASK (0xF << 0)
+#define DA7219_DAC_EQ_BAND4_SHIFT 4
+#define DA7219_DAC_EQ_BAND4_MASK (0xF << 4)
+
+/* DA7219_DAC_FILTERS4 = 0x43 */
+#define DA7219_DAC_EQ_BAND5_SHIFT 0
+#define DA7219_DAC_EQ_BAND5_MASK (0xF << 0)
+#define DA7219_DAC_EQ_EN_SHIFT 7
+#define DA7219_DAC_EQ_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_FILTERS1 = 0x44 */
+#define DA7219_DAC_VOICE_HPF_CORNER_SHIFT 0
+#define DA7219_DAC_VOICE_HPF_CORNER_MASK (0x7 << 0)
+#define DA7219_DAC_VOICE_EN_SHIFT 3
+#define DA7219_DAC_VOICE_EN_MASK (0x1 << 3)
+#define DA7219_DAC_AUDIO_HPF_CORNER_SHIFT 4
+#define DA7219_DAC_AUDIO_HPF_CORNER_MASK (0x3 << 4)
+#define DA7219_DAC_HPF_EN_SHIFT 7
+#define DA7219_DAC_HPF_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_L_GAIN = 0x45 */
+#define DA7219_DAC_L_DIGITAL_GAIN_SHIFT 0
+#define DA7219_DAC_L_DIGITAL_GAIN_MASK (0x7F << 0)
+#define DA7219_DAC_DIGITAL_GAIN_MAX 0x7F
+#define DA7219_DAC_DIGITAL_GAIN_0DB (0x6F << 0)
+
+/* DA7219_DAC_R_GAIN = 0x46 */
+#define DA7219_DAC_R_DIGITAL_GAIN_SHIFT 0
+#define DA7219_DAC_R_DIGITAL_GAIN_MASK (0x7F << 0)
+
+/* DA7219_CP_CTRL = 0x47 */
+#define DA7219_CP_MCHANGE_SHIFT 4
+#define DA7219_CP_MCHANGE_MASK (0x3 << 4)
+#define DA7219_CP_MCHANGE_REL_MASK 0x3
+#define DA7219_CP_MCHANGE_MAX 3
+#define DA7219_CP_MCHANGE_LARGEST_VOL 0x1
+#define DA7219_CP_MCHANGE_DAC_VOL 0x2
+#define DA7219_CP_MCHANGE_SIG_MAG 0x3
+#define DA7219_CP_EN_SHIFT 7
+#define DA7219_CP_EN_MASK (0x1 << 7)
+
+/* DA7219_HP_L_GAIN = 0x48 */
+#define DA7219_HP_L_AMP_GAIN_SHIFT 0
+#define DA7219_HP_L_AMP_GAIN_MASK (0x3F << 0)
+#define DA7219_HP_AMP_GAIN_MAX 0x3F
+#define DA7219_HP_AMP_GAIN_0DB (0x39 << 0)
+
+/* DA7219_HP_R_GAIN = 0x49 */
+#define DA7219_HP_R_AMP_GAIN_SHIFT 0
+#define DA7219_HP_R_AMP_GAIN_MASK (0x3F << 0)
+
+/* DA7219_MIXOUT_L_SELECT = 0x4B */
+#define DA7219_MIXOUT_L_MIX_SELECT_SHIFT 0
+#define DA7219_MIXOUT_L_MIX_SELECT_MASK (0x1 << 0)
+
+/* DA7219_MIXOUT_R_SELECT = 0x4C */
+#define DA7219_MIXOUT_R_MIX_SELECT_SHIFT 0
+#define DA7219_MIXOUT_R_MIX_SELECT_MASK (0x1 << 0)
+
+/* DA7219_SYSTEM_MODES_INPUT = 0x50 */
+#define DA7219_MODE_SUBMIT_SHIFT 0
+#define DA7219_MODE_SUBMIT_MASK (0x1 << 0)
+#define DA7219_ADC_MODE_SHIFT 1
+#define DA7219_ADC_MODE_MASK (0x7F << 1)
+
+/* DA7219_SYSTEM_MODES_OUTPUT = 0x51 */
+#define DA7219_MODE_SUBMIT_SHIFT 0
+#define DA7219_MODE_SUBMIT_MASK (0x1 << 0)
+#define DA7219_DAC_MODE_SHIFT 1
+#define DA7219_DAC_MODE_MASK (0x7F << 1)
+
+/* DA7219_MICBIAS_CTRL = 0x62 */
+#define DA7219_MICBIAS1_LEVEL_SHIFT 0
+#define DA7219_MICBIAS1_LEVEL_MASK (0x7 << 0)
+#define DA7219_MICBIAS1_EN_SHIFT 3
+#define DA7219_MICBIAS1_EN_MASK (0x1 << 3)
+
+/* DA7219_MIC_1_CTRL = 0x63 */
+#define DA7219_MIC_1_AMP_RAMP_EN_SHIFT 5
+#define DA7219_MIC_1_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_MIC_1_AMP_MUTE_EN_SHIFT 6
+#define DA7219_MIC_1_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_MIC_1_AMP_EN_SHIFT 7
+#define DA7219_MIC_1_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_MIXIN_L_CTRL = 0x65 */
+#define DA7219_MIXIN_L_MIX_EN_SHIFT 3
+#define DA7219_MIXIN_L_MIX_EN_MASK (0x1 << 3)
+#define DA7219_MIXIN_L_AMP_ZC_EN_SHIFT 4
+#define DA7219_MIXIN_L_AMP_ZC_EN_MASK (0x1 << 4)
+#define DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT 5
+#define DA7219_MIXIN_L_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT 6
+#define DA7219_MIXIN_L_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_MIXIN_L_AMP_EN_SHIFT 7
+#define DA7219_MIXIN_L_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_ADC_L_CTRL = 0x67 */
+#define DA7219_ADC_L_BIAS_SHIFT 0
+#define DA7219_ADC_L_BIAS_MASK (0x3 << 0)
+#define DA7219_ADC_L_RAMP_EN_SHIFT 5
+#define DA7219_ADC_L_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_ADC_L_MUTE_EN_SHIFT 6
+#define DA7219_ADC_L_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_ADC_L_EN_SHIFT 7
+#define DA7219_ADC_L_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_L_CTRL = 0x69 */
+#define DA7219_DAC_L_RAMP_EN_SHIFT 5
+#define DA7219_DAC_L_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_DAC_L_MUTE_EN_SHIFT 6
+#define DA7219_DAC_L_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_DAC_L_EN_SHIFT 7
+#define DA7219_DAC_L_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_R_CTRL = 0x6A */
+#define DA7219_DAC_R_RAMP_EN_SHIFT 5
+#define DA7219_DAC_R_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_DAC_R_MUTE_EN_SHIFT 6
+#define DA7219_DAC_R_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_DAC_R_EN_SHIFT 7
+#define DA7219_DAC_R_EN_MASK (0x1 << 7)
+
+/* DA7219_HP_L_CTRL = 0x6B */
+#define DA7219_HP_L_AMP_MIN_GAIN_EN_SHIFT 2
+#define DA7219_HP_L_AMP_MIN_GAIN_EN_MASK (0x1 << 2)
+#define DA7219_HP_L_AMP_OE_SHIFT 3
+#define DA7219_HP_L_AMP_OE_MASK (0x1 << 3)
+#define DA7219_HP_L_AMP_ZC_EN_SHIFT 4
+#define DA7219_HP_L_AMP_ZC_EN_MASK (0x1 << 4)
+#define DA7219_HP_L_AMP_RAMP_EN_SHIFT 5
+#define DA7219_HP_L_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_HP_L_AMP_MUTE_EN_SHIFT 6
+#define DA7219_HP_L_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_HP_L_AMP_EN_SHIFT 7
+#define DA7219_HP_L_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_HP_R_CTRL = 0x6C */
+#define DA7219_HP_R_AMP_MIN_GAIN_EN_SHIFT 2
+#define DA7219_HP_R_AMP_MIN_GAIN_EN_MASK (0x1 << 2)
+#define DA7219_HP_R_AMP_OE_SHIFT 3
+#define DA7219_HP_R_AMP_OE_MASK (0x1 << 3)
+#define DA7219_HP_R_AMP_ZC_EN_SHIFT 4
+#define DA7219_HP_R_AMP_ZC_EN_MASK (0x1 << 4)
+#define DA7219_HP_R_AMP_RAMP_EN_SHIFT 5
+#define DA7219_HP_R_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_HP_R_AMP_MUTE_EN_SHIFT 6
+#define DA7219_HP_R_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_HP_R_AMP_EN_SHIFT 7
+#define DA7219_HP_R_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_MIXOUT_L_CTRL = 0x6E */
+#define DA7219_MIXOUT_L_AMP_EN_SHIFT 7
+#define DA7219_MIXOUT_L_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_MIXOUT_R_CTRL = 0x6F */
+#define DA7219_MIXOUT_R_AMP_EN_SHIFT 7
+#define DA7219_MIXOUT_R_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_CHIP_ID1 = 0x81 */
+#define DA7219_CHIP_ID1_SHIFT 0
+#define DA7219_CHIP_ID1_MASK (0xFF << 0)
+
+/* DA7219_CHIP_ID2 = 0x82 */
+#define DA7219_CHIP_ID2_SHIFT 0
+#define DA7219_CHIP_ID2_MASK (0xFF << 0)
+
+/* DA7219_CHIP_REVISION = 0x83 */
+#define DA7219_CHIP_MINOR_SHIFT 0
+#define DA7219_CHIP_MINOR_MASK (0xF << 0)
+#define DA7219_CHIP_MAJOR_SHIFT 4
+#define DA7219_CHIP_MAJOR_MASK (0xF << 4)
+
+/* DA7219_LDO_CTRL = 0x90 */
+#define DA7219_LDO_LEVEL_SELECT_SHIFT 4
+#define DA7219_LDO_LEVEL_SELECT_MASK (0x3 << 4)
+#define DA7219_LDO_EN_SHIFT 7
+#define DA7219_LDO_EN_MASK (0x1 << 7)
+
+/* DA7219_IO_CTRL = 0x91 */
+#define DA7219_IO_VOLTAGE_LEVEL_SHIFT 0
+#define DA7219_IO_VOLTAGE_LEVEL_MASK (0x1 << 0)
+#define DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V 0
+#define DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V 1
+
+/* DA7219_GAIN_RAMP_CTRL = 0x92 */
+#define DA7219_GAIN_RAMP_RATE_SHIFT 0
+#define DA7219_GAIN_RAMP_RATE_MASK (0x3 << 0)
+#define DA7219_GAIN_RAMP_RATE_MAX 4
+
+/* DA7219_PC_COUNT = 0x94 */
+#define DA7219_PC_FREERUN_SHIFT 0
+#define DA7219_PC_FREERUN_MASK (0x1 << 0)
+#define DA7219_PC_RESYNC_AUTO_SHIFT 1
+#define DA7219_PC_RESYNC_AUTO_MASK (0x1 << 1)
+
+/* DA7219_CP_VOL_THRESHOLD1 = 0x95 */
+#define DA7219_CP_THRESH_VDD2_SHIFT 0
+#define DA7219_CP_THRESH_VDD2_MASK (0x3F << 0)
+#define DA7219_CP_THRESH_VDD2_MAX 0x3F
+
+/* DA7219_DIG_CTRL = 0x99 */
+#define DA7219_DAC_L_INV_SHIFT 3
+#define DA7219_DAC_L_INV_MASK (0x1 << 3)
+#define DA7219_DAC_R_INV_SHIFT 7
+#define DA7219_DAC_R_INV_MASK (0x1 << 7)
+
+/* DA7219_ALC_CTRL2 = 0x9A */
+#define DA7219_ALC_ATTACK_SHIFT 0
+#define DA7219_ALC_ATTACK_MASK (0xF << 0)
+#define DA7219_ALC_ATTACK_MAX 13
+#define DA7219_ALC_RELEASE_SHIFT 4
+#define DA7219_ALC_RELEASE_MASK (0xF << 4)
+#define DA7219_ALC_RELEASE_MAX 11
+
+/* DA7219_ALC_CTRL3 = 0x9B */
+#define DA7219_ALC_HOLD_SHIFT 0
+#define DA7219_ALC_HOLD_MASK (0xF << 0)
+#define DA7219_ALC_HOLD_MAX 16
+#define DA7219_ALC_INTEG_ATTACK_SHIFT 4
+#define DA7219_ALC_INTEG_ATTACK_MASK (0x3 << 4)
+#define DA7219_ALC_INTEG_RELEASE_SHIFT 6
+#define DA7219_ALC_INTEG_RELEASE_MASK (0x3 << 6)
+#define DA7219_ALC_INTEG_MAX 4
+
+/* DA7219_ALC_NOISE = 0x9C */
+#define DA7219_ALC_NOISE_SHIFT 0
+#define DA7219_ALC_NOISE_MASK (0x3F << 0)
+#define DA7219_ALC_THRESHOLD_MAX 0x3F
+
+/* DA7219_ALC_TARGET_MIN = 0x9D */
+#define DA7219_ALC_THRESHOLD_MIN_SHIFT 0
+#define DA7219_ALC_THRESHOLD_MIN_MASK (0x3F << 0)
+
+/* DA7219_ALC_TARGET_MAX = 0x9E */
+#define DA7219_ALC_THRESHOLD_MAX_SHIFT 0
+#define DA7219_ALC_THRESHOLD_MAX_MASK (0x3F << 0)
+
+/* DA7219_ALC_GAIN_LIMITS = 0x9F */
+#define DA7219_ALC_ATTEN_MAX_SHIFT 0
+#define DA7219_ALC_ATTEN_MAX_MASK (0xF << 0)
+#define DA7219_ALC_GAIN_MAX_SHIFT 4
+#define DA7219_ALC_GAIN_MAX_MASK (0xF << 4)
+#define DA7219_ALC_ATTEN_GAIN_MAX 0xF
+
+/* DA7219_ALC_ANA_GAIN_LIMITS = 0xA0 */
+#define DA7219_ALC_ANA_GAIN_MIN_SHIFT 0
+#define DA7219_ALC_ANA_GAIN_MIN_MASK (0x7 << 0)
+#define DA7219_ALC_ANA_GAIN_MIN 0x1
+#define DA7219_ALC_ANA_GAIN_MAX_SHIFT 4
+#define DA7219_ALC_ANA_GAIN_MAX_MASK (0x7 << 4)
+#define DA7219_ALC_ANA_GAIN_MAX 0x7
+
+/* DA7219_ALC_ANTICLIP_CTRL = 0xA1 */
+#define DA7219_ALC_ANTICLIP_STEP_SHIFT 0
+#define DA7219_ALC_ANTICLIP_STEP_MASK (0x3 << 0)
+#define DA7219_ALC_ANTICLIP_STEP_MAX 4
+#define DA7219_ALC_ANTIPCLIP_EN_SHIFT 7
+#define DA7219_ALC_ANTIPCLIP_EN_MASK (0x1 << 7)
+
+/* DA7219_ALC_ANTICLIP_LEVEL = 0xA2 */
+#define DA7219_ALC_ANTICLIP_LEVEL_SHIFT 0
+#define DA7219_ALC_ANTICLIP_LEVEL_MASK (0x7F << 0)
+
+/* DA7219_ALC_OFFSET_AUTO_M_L = 0xA3 */
+#define DA7219_ALC_OFFSET_AUTO_M_L_SHIFT 0
+#define DA7219_ALC_OFFSET_AUTO_M_L_MASK (0xFF << 0)
+
+/* DA7219_ALC_OFFSET_AUTO_U_L = 0xA4 */
+#define DA7219_ALC_OFFSET_AUTO_U_L_SHIFT 0
+#define DA7219_ALC_OFFSET_AUTO_U_L_MASK (0xF << 0)
+
+/* DA7219_DAC_NG_SETUP_TIME = 0xAF */
+#define DA7219_DAC_NG_SETUP_TIME_SHIFT 0
+#define DA7219_DAC_NG_SETUP_TIME_MASK (0x3 << 0)
+#define DA7219_DAC_NG_SETUP_TIME_MAX 4
+#define DA7219_DAC_NG_RAMPUP_RATE_SHIFT 2
+#define DA7219_DAC_NG_RAMPUP_RATE_MASK (0x1 << 2)
+#define DA7219_DAC_NG_RAMPDN_RATE_SHIFT 3
+#define DA7219_DAC_NG_RAMPDN_RATE_MASK (0x1 << 3)
+#define DA7219_DAC_NG_RAMP_RATE_MAX 2
+
+/* DA7219_DAC_NG_OFF_THRESH = 0xB0 */
+#define DA7219_DAC_NG_OFF_THRESHOLD_SHIFT 0
+#define DA7219_DAC_NG_OFF_THRESHOLD_MASK (0x7 << 0)
+#define DA7219_DAC_NG_THRESHOLD_MAX 0x7
+
+/* DA7219_DAC_NG_ON_THRESH = 0xB1 */
+#define DA7219_DAC_NG_ON_THRESHOLD_SHIFT 0
+#define DA7219_DAC_NG_ON_THRESHOLD_MASK (0x7 << 0)
+
+/* DA7219_DAC_NG_CTRL = 0xB2 */
+#define DA7219_DAC_NG_EN_SHIFT 7
+#define DA7219_DAC_NG_EN_MASK (0x1 << 7)
+
+/* DA7219_TONE_GEN_CFG1 = 0xB4 */
+#define DA7219_DTMF_REG_SHIFT 0
+#define DA7219_DTMF_REG_MASK (0xF << 0)
+#define DA7219_DTMF_REG_MAX 16
+#define DA7219_DTMF_EN_SHIFT 4
+#define DA7219_DTMF_EN_MASK (0x1 << 4)
+#define DA7219_START_STOPN_SHIFT 7
+#define DA7219_START_STOPN_MASK (0x1 << 7)
+
+/* DA7219_TONE_GEN_CFG2 = 0xB5 */
+#define DA7219_SWG_SEL_SHIFT 0
+#define DA7219_SWG_SEL_MASK (0x3 << 0)
+#define DA7219_SWG_SEL_MAX 4
+#define DA7219_SWG_SEL_SRAMP (0x3 << 0)
+#define DA7219_TONE_GEN_GAIN_SHIFT 4
+#define DA7219_TONE_GEN_GAIN_MASK (0xF << 4)
+#define DA7219_TONE_GEN_GAIN_MAX 0xF
+#define DA7219_TONE_GEN_GAIN_MINUS_9DB (0x3 << 4)
+#define DA7219_TONE_GEN_GAIN_MINUS_15DB (0x5 << 4)
+
+/* DA7219_TONE_GEN_CYCLES = 0xB6 */
+#define DA7219_BEEP_CYCLES_SHIFT 0
+#define DA7219_BEEP_CYCLES_MASK (0x7 << 0)
+
+/* DA7219_TONE_GEN_FREQ1_L = 0xB7 */
+#define DA7219_FREQ1_L_SHIFT 0
+#define DA7219_FREQ1_L_MASK (0xFF << 0)
+#define DA7219_FREQ_MAX 0xFFFF
+
+/* DA7219_TONE_GEN_FREQ1_U = 0xB8 */
+#define DA7219_FREQ1_U_SHIFT 0
+#define DA7219_FREQ1_U_MASK (0xFF << 0)
+
+/* DA7219_TONE_GEN_FREQ2_L = 0xB9 */
+#define DA7219_FREQ2_L_SHIFT 0
+#define DA7219_FREQ2_L_MASK (0xFF << 0)
+
+/* DA7219_TONE_GEN_FREQ2_U = 0xBA */
+#define DA7219_FREQ2_U_SHIFT 0
+#define DA7219_FREQ2_U_MASK (0xFF << 0)
+
+/* DA7219_TONE_GEN_ON_PER = 0xBB */
+#define DA7219_BEEP_ON_PER_SHIFT 0
+#define DA7219_BEEP_ON_PER_MASK (0x3F << 0)
+#define DA7219_BEEP_ON_OFF_MAX 0x3F
+
+/* DA7219_TONE_GEN_OFF_PER = 0xBC */
+#define DA7219_BEEP_OFF_PER_SHIFT 0
+#define DA7219_BEEP_OFF_PER_MASK (0x3F << 0)
+
+/* DA7219_SYSTEM_STATUS = 0xE0 */
+#define DA7219_SC1_BUSY_SHIFT 0
+#define DA7219_SC1_BUSY_MASK (0x1 << 0)
+#define DA7219_SC2_BUSY_SHIFT 1
+#define DA7219_SC2_BUSY_MASK (0x1 << 1)
+
+/* DA7219_SYSTEM_ACTIVE = 0xFD */
+#define DA7219_SYSTEM_ACTIVE_SHIFT 0
+#define DA7219_SYSTEM_ACTIVE_MASK (0x1 << 0)
+
+
+/*
+ * General defines & data
+ */
+
+/* Register inversion */
+#define DA7219_NO_INVERT 0
+#define DA7219_INVERT 1
+
+/* Byte related defines */
+#define DA7219_BYTE_SHIFT 8
+#define DA7219_BYTE_MASK 0xFF
+
+/* PLL Output Frequencies */
+#define DA7219_PLL_FREQ_OUT_90316 90316800
+#define DA7219_PLL_FREQ_OUT_98304 98304000
+
+/* PLL Frequency Dividers */
+#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1
+#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2
+#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4
+#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8
+#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16
+
+/* SRM */
+#define DA7219_SRM_CHECK_RETRIES 8
+
+enum da7219_clk_src {
+ DA7219_CLKSRC_MCLK = 0,
+ DA7219_CLKSRC_MCLK_SQR,
+};
+
+enum da7219_sys_clk {
+ DA7219_SYSCLK_MCLK = 0,
+ DA7219_SYSCLK_PLL,
+ DA7219_SYSCLK_PLL_SRM,
+ DA7219_SYSCLK_PLL_32KHZ
+};
+
+/* Regulators */
+enum da7219_supplies {
+ DA7219_SUPPLY_VDD = 0,
+ DA7219_SUPPLY_VDDMIC,
+ DA7219_SUPPLY_VDDIO,
+ DA7219_NUM_SUPPLIES,
+};
+
+struct da7219_aad_priv;
+
+/* Private data */
+struct da7219_priv {
+ struct da7219_aad_priv *aad;
+ struct da7219_pdata *pdata;
+
+ struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES];
+ struct regmap *regmap;
+ struct mutex lock;
+
+ struct clk *mclk;
+ unsigned int mclk_rate;
+ int clk_src;
+
+ bool master;
+ bool alc_en;
+};
+
+#endif /* __DA7219_H */
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 6a09101..969e337 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -129,7 +129,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
- int deemph = ucontrol->value.integer.value[0];
+ unsigned int deemph = ucontrol->value.integer.value[0];
int ret;
if (deemph > 1)
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
deleted file mode 100644
index bd42ad3..0000000
--- a/sound/soc/codecs/hdmi.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * ALSA SoC codec driver for HDMI audio codecs.
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <linux/of.h>
-#include <linux/of_device.h>
-
-#define DRV_NAME "hdmi-audio-codec"
-
-static const struct snd_soc_dapm_widget hdmi_widgets[] = {
- SND_SOC_DAPM_INPUT("RX"),
- SND_SOC_DAPM_OUTPUT("TX"),
-};
-
-static const struct snd_soc_dapm_route hdmi_routes[] = {
- { "Capture", NULL, "RX" },
- { "TX", NULL, "Playback" },
-};
-
-static struct snd_soc_dai_driver hdmi_codec_dai = {
- .name = "hdmi-hifi",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 2,
- .channels_max = 8,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
- .sig_bits = 24,
- },
- .capture = {
- .stream_name = "Capture",
- .channels_min = 2,
- .channels_max = 2,
- .rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE,
- },
-
-};
-
-#ifdef CONFIG_OF
-static const struct of_device_id hdmi_audio_codec_ids[] = {
- { .compatible = "linux,hdmi-audio", },
- { }
-};
-MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids);
-#endif
-
-static struct snd_soc_codec_driver hdmi_codec = {
- .dapm_widgets = hdmi_widgets,
- .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
- .dapm_routes = hdmi_routes,
- .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
- .ignore_pmdown_time = true,
-};
-
-static int hdmi_codec_probe(struct platform_device *pdev)
-{
- return snd_soc_register_codec(&pdev->dev, &hdmi_codec,
- &hdmi_codec_dai, 1);
-}
-
-static int hdmi_codec_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_codec(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver hdmi_codec_driver = {
- .driver = {
- .name = DRV_NAME,
- .of_match_table = of_match_ptr(hdmi_audio_codec_ids),
- },
-
- .probe = hdmi_codec_probe,
- .remove = hdmi_codec_remove,
-};
-
-module_platform_driver(hdmi_codec_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("ASoC generic HDMI codec driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
new file mode 100644
index 0000000..7fc7b4e
--- /dev/null
+++ b/sound/soc/codecs/nau8825.c
@@ -0,0 +1,1309 @@
+/*
+ * Nuvoton NAU8825 audio codec driver
+ *
+ * Copyright 2015 Google Chromium project.
+ * Author: Anatol Pomozov <anatol@chromium.org>
+ * Copyright 2015 Nuvoton Technology Corp.
+ * Co-author: Meng-Huang Kuo <mhkuo@nuvoton.com>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <linux/acpi.h>
+#include <linux/math64.h>
+
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+
+#include "nau8825.h"
+
+#define NAU_FREF_MAX 13500000
+#define NAU_FVCO_MAX 100000000
+#define NAU_FVCO_MIN 90000000
+
+struct nau8825_fll {
+ int mclk_src;
+ int ratio;
+ int fll_frac;
+ int fll_int;
+ int clk_ref_div;
+};
+
+struct nau8825_fll_attr {
+ unsigned int param;
+ unsigned int val;
+};
+
+/* scaling for mclk from sysclk_src output */
+static const struct nau8825_fll_attr mclk_src_scaling[] = {
+ { 1, 0x0 },
+ { 2, 0x2 },
+ { 4, 0x3 },
+ { 8, 0x4 },
+ { 16, 0x5 },
+ { 32, 0x6 },
+ { 3, 0x7 },
+ { 6, 0xa },
+ { 12, 0xb },
+ { 24, 0xc },
+ { 48, 0xd },
+ { 96, 0xe },
+ { 5, 0xf },
+};
+
+/* ratio for input clk freq */
+static const struct nau8825_fll_attr fll_ratio[] = {
+ { 512000, 0x01 },
+ { 256000, 0x02 },
+ { 128000, 0x04 },
+ { 64000, 0x08 },
+ { 32000, 0x10 },
+ { 8000, 0x20 },
+ { 4000, 0x40 },
+};
+
+static const struct nau8825_fll_attr fll_pre_scalar[] = {
+ { 1, 0x0 },
+ { 2, 0x1 },
+ { 4, 0x2 },
+ { 8, 0x3 },
+};
+
+static const struct reg_default nau8825_reg_defaults[] = {
+ { NAU8825_REG_ENA_CTRL, 0x00ff },
+ { NAU8825_REG_CLK_DIVIDER, 0x0050 },
+ { NAU8825_REG_FLL1, 0x0 },
+ { NAU8825_REG_FLL2, 0x3126 },
+ { NAU8825_REG_FLL3, 0x0008 },
+ { NAU8825_REG_FLL4, 0x0010 },
+ { NAU8825_REG_FLL5, 0x0 },
+ { NAU8825_REG_FLL6, 0x6000 },
+ { NAU8825_REG_FLL_VCO_RSV, 0xf13c },
+ { NAU8825_REG_HSD_CTRL, 0x000c },
+ { NAU8825_REG_JACK_DET_CTRL, 0x0 },
+ { NAU8825_REG_INTERRUPT_MASK, 0x0 },
+ { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff },
+ { NAU8825_REG_SAR_CTRL, 0x0015 },
+ { NAU8825_REG_KEYDET_CTRL, 0x0110 },
+ { NAU8825_REG_VDET_THRESHOLD_1, 0x0 },
+ { NAU8825_REG_VDET_THRESHOLD_2, 0x0 },
+ { NAU8825_REG_VDET_THRESHOLD_3, 0x0 },
+ { NAU8825_REG_VDET_THRESHOLD_4, 0x0 },
+ { NAU8825_REG_GPIO34_CTRL, 0x0 },
+ { NAU8825_REG_GPIO12_CTRL, 0x0 },
+ { NAU8825_REG_TDM_CTRL, 0x0 },
+ { NAU8825_REG_I2S_PCM_CTRL1, 0x000b },
+ { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 },
+ { NAU8825_REG_LEFT_TIME_SLOT, 0x0 },
+ { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 },
+ { NAU8825_REG_BIQ_CTRL, 0x0 },
+ { NAU8825_REG_BIQ_COF1, 0x0 },
+ { NAU8825_REG_BIQ_COF2, 0x0 },
+ { NAU8825_REG_BIQ_COF3, 0x0 },
+ { NAU8825_REG_BIQ_COF4, 0x0 },
+ { NAU8825_REG_BIQ_COF5, 0x0 },
+ { NAU8825_REG_BIQ_COF6, 0x0 },
+ { NAU8825_REG_BIQ_COF7, 0x0 },
+ { NAU8825_REG_BIQ_COF8, 0x0 },
+ { NAU8825_REG_BIQ_COF9, 0x0 },
+ { NAU8825_REG_BIQ_COF10, 0x0 },
+ { NAU8825_REG_ADC_RATE, 0x0010 },
+ { NAU8825_REG_DAC_CTRL1, 0x0001 },
+ { NAU8825_REG_DAC_CTRL2, 0x0 },
+ { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 },
+ { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf },
+ { NAU8825_REG_MUTE_CTRL, 0x0 },
+ { NAU8825_REG_HSVOL_CTRL, 0x0 },
+ { NAU8825_REG_DACL_CTRL, 0x02cf },
+ { NAU8825_REG_DACR_CTRL, 0x00cf },
+ { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 },
+ { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 },
+ { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff },
+ { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 },
+ { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 },
+ { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 },
+ { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 },
+ { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 },
+ { NAU8825_REG_IMM_MODE_CTRL, 0x0 },
+ { NAU8825_REG_CLASSG_CTRL, 0x0 },
+ { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 },
+ { NAU8825_REG_MISC_CTRL, 0x0 },
+ { NAU8825_REG_BIAS_ADJ, 0x0 },
+ { NAU8825_REG_TRIM_SETTINGS, 0x0 },
+ { NAU8825_REG_ANALOG_CONTROL_1, 0x0 },
+ { NAU8825_REG_ANALOG_CONTROL_2, 0x0 },
+ { NAU8825_REG_ANALOG_ADC_1, 0x0011 },
+ { NAU8825_REG_ANALOG_ADC_2, 0x0020 },
+ { NAU8825_REG_RDAC, 0x0008 },
+ { NAU8825_REG_MIC_BIAS, 0x0006 },
+ { NAU8825_REG_BOOST, 0x0 },
+ { NAU8825_REG_FEPGA, 0x0 },
+ { NAU8825_REG_POWER_UP_CONTROL, 0x0 },
+ { NAU8825_REG_CHARGE_PUMP, 0x0 },
+};
+
+static bool nau8825_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8825_REG_ENA_CTRL:
+ case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV:
+ case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL:
+ case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL:
+ case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL:
+ case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY:
+ case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY:
+ case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R:
+ case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL:
+ case NAU8825_REG_MISC_CTRL:
+ case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS:
+ case NAU8825_REG_BIAS_ADJ:
+ case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2:
+ case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS:
+ case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA:
+ case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS:
+ return true;
+ default:
+ return false;
+ }
+
+}
+
+static bool nau8825_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL:
+ case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV:
+ case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL:
+ case NAU8825_REG_INTERRUPT_MASK:
+ case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL:
+ case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL:
+ case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY:
+ case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY:
+ case NAU8825_REG_IMM_MODE_CTRL:
+ case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL:
+ case NAU8825_REG_MISC_CTRL:
+ case NAU8825_REG_BIAS_ADJ:
+ case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2:
+ case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS:
+ case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA:
+ case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool nau8825_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case NAU8825_REG_RESET:
+ case NAU8825_REG_IRQ_STATUS:
+ case NAU8825_REG_INT_CLR_KEY_STATUS:
+ case NAU8825_REG_IMM_RMS_L:
+ case NAU8825_REG_IMM_RMS_R:
+ case NAU8825_REG_I2C_DEVICE_ID:
+ case NAU8825_REG_SARDOUT_RAM_STATUS:
+ case NAU8825_REG_CHARGE_PUMP_INPUT_READ:
+ case NAU8825_REG_GENERAL_STATUS:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static int nau8825_pump_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* Prevent startup click by letting charge pump to ramp up */
+ msleep(10);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const char * const nau8825_adc_decimation[] = {
+ "32", "64", "128", "256"
+};
+
+static const struct soc_enum nau8825_adc_decimation_enum =
+ SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT,
+ ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation);
+
+static const char * const nau8825_dac_oversampl[] = {
+ "64", "256", "128", "", "32"
+};
+
+static const struct soc_enum nau8825_dac_oversampl_enum =
+ SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT,
+ ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl);
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400);
+static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0);
+static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600);
+static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -9600, 2400);
+
+static const struct snd_kcontrol_new nau8825_controls[] = {
+ SOC_SINGLE_TLV("Mic Volume", NAU8825_REG_ADC_DGAIN_CTRL,
+ 0, 0xff, 0, adc_vol_tlv),
+ SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8825_REG_ADC_DGAIN_CTRL,
+ 12, 8, 0x0f, 0, sidetone_vol_tlv),
+ SOC_DOUBLE_TLV("Headphone Volume", NAU8825_REG_HSVOL_CTRL,
+ 6, 0, 0x3f, 1, dac_vol_tlv),
+ SOC_SINGLE_TLV("Frontend PGA Volume", NAU8825_REG_POWER_UP_CONTROL,
+ 8, 37, 0, fepga_gain_tlv),
+ SOC_DOUBLE_TLV("Headphone Crosstalk Volume", NAU8825_REG_DAC_DGAIN_CTRL,
+ 0, 8, 0xff, 0, crosstalk_vol_tlv),
+
+ SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum),
+ SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum),
+};
+
+/* DAC Mux 0x33[9] and 0x34[9] */
+static const char * const nau8825_dac_src[] = {
+ "DACL", "DACR",
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ nau8825_dacl_enum, NAU8825_REG_DACL_CTRL,
+ NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src);
+
+static SOC_ENUM_SINGLE_DECL(
+ nau8825_dacr_enum, NAU8825_REG_DACR_CTRL,
+ NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src);
+
+static const struct snd_kcontrol_new nau8825_dacl_mux =
+ SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum);
+
+static const struct snd_kcontrol_new nau8825_dacr_mux =
+ SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum);
+
+
+static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2,
+ 15, 1),
+
+ SND_SOC_DAPM_INPUT("MIC"),
+ SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0),
+
+ SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0),
+ SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL,
+ 0),
+
+ /* ADC for button press detection */
+ SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL,
+ NAU8825_SAR_ADC_EN_SFT, 0),
+
+ SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0),
+ SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0),
+ SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0),
+
+ SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL,
+ NAU8825_ENABLE_DACR_SFT, 0),
+ SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL,
+ NAU8825_ENABLE_DACL_SFT, 0),
+ SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux),
+ SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux),
+
+ SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL,
+ 0),
+
+ SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0,
+ nau8825_pump_event, SND_SOC_DAPM_POST_PMU),
+
+ SND_SOC_DAPM_PGA("Output Driver R Stage 1",
+ NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Output Driver L Stage 1",
+ NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Output Driver R Stage 2",
+ NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Output Driver L Stage 2",
+ NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1,
+ NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1,
+ NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0),
+ SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("HPOL"),
+ SND_SOC_DAPM_OUTPUT("HPOR"),
+};
+
+static const struct snd_soc_dapm_route nau8825_dapm_routes[] = {
+ {"Frontend PGA", NULL, "MIC"},
+ {"ADC", NULL, "Frontend PGA"},
+ {"ADC", NULL, "ADC Clock"},
+ {"ADC", NULL, "ADC Power"},
+ {"AIFTX", NULL, "ADC"},
+
+ {"DDACL", NULL, "Playback"},
+ {"DDACR", NULL, "Playback"},
+ {"DDACL", NULL, "DDAC Clock"},
+ {"DDACR", NULL, "DDAC Clock"},
+ {"DACL Mux", "DACL", "DDACL"},
+ {"DACL Mux", "DACR", "DDACR"},
+ {"DACR Mux", "DACL", "DDACL"},
+ {"DACR Mux", "DACR", "DDACR"},
+ {"HP amp L", NULL, "DACL Mux"},
+ {"HP amp R", NULL, "DACR Mux"},
+ {"HP amp L", NULL, "HP amp power"},
+ {"HP amp R", NULL, "HP amp power"},
+ {"ADACL", NULL, "HP amp L"},
+ {"ADACR", NULL, "HP amp R"},
+ {"ADACL", NULL, "ADACL Clock"},
+ {"ADACR", NULL, "ADACR Clock"},
+ {"Output Driver L Stage 1", NULL, "ADACL"},
+ {"Output Driver R Stage 1", NULL, "ADACR"},
+ {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"},
+ {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"},
+ {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"},
+ {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"},
+ {"Output DACL", NULL, "Output Driver L Stage 3"},
+ {"Output DACR", NULL, "Output Driver R Stage 3"},
+ {"HPOL", NULL, "Output DACL"},
+ {"HPOR", NULL, "Output DACR"},
+ {"HPOL", NULL, "Charge Pump"},
+ {"HPOR", NULL, "Charge Pump"},
+};
+
+static int nau8825_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val_len = 0;
+
+ switch (params_width(params)) {
+ case 16:
+ val_len |= NAU8825_I2S_DL_16;
+ break;
+ case 20:
+ val_len |= NAU8825_I2S_DL_20;
+ break;
+ case 24:
+ val_len |= NAU8825_I2S_DL_24;
+ break;
+ case 32:
+ val_len |= NAU8825_I2S_DL_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1,
+ NAU8825_I2S_DL_MASK, val_len);
+
+ return 0;
+}
+
+static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+ unsigned int ctrl1_val = 0, ctrl2_val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ctrl2_val |= NAU8825_I2S_MS_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ ctrl1_val |= NAU8825_I2S_BP_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ctrl1_val |= NAU8825_I2S_DF_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ctrl1_val |= NAU8825_I2S_DF_LEFT;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ ctrl1_val |= NAU8825_I2S_DF_RIGTH;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ ctrl1_val |= NAU8825_I2S_DF_PCM_AB;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ ctrl1_val |= NAU8825_I2S_DF_PCM_AB;
+ ctrl1_val |= NAU8825_I2S_PCMB_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1,
+ NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK |
+ NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK,
+ ctrl1_val);
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2,
+ NAU8825_I2S_MS_MASK, ctrl2_val);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops nau8825_dai_ops = {
+ .hw_params = nau8825_hw_params,
+ .set_fmt = nau8825_set_dai_fmt,
+};
+
+#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000
+#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver nau8825_dai = {
+ .name = "nau8825-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = NAU8825_RATES,
+ .formats = NAU8825_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = NAU8825_RATES,
+ .formats = NAU8825_FORMATS,
+ },
+ .ops = &nau8825_dai_ops,
+};
+
+/**
+ * nau8825_enable_jack_detect - Specify a jack for event reporting
+ *
+ * @component: component to register the jack with
+ * @jack: jack to use to report headset and button events on
+ *
+ * After this function has been called the headset insert/remove and button
+ * events will be routed to the given jack. Jack can be null to stop
+ * reporting.
+ */
+int nau8825_enable_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *jack)
+{
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+ struct regmap *regmap = nau8825->regmap;
+
+ nau8825->jack = jack;
+
+ /* Ground HP Outputs[1:0], needed for headset auto detection
+ * Enable Automatic Mic/Gnd switching reading on insert interrupt[6]
+ */
+ regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL,
+ NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L,
+ NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L);
+
+ regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+ NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect);
+
+
+static bool nau8825_is_jack_inserted(struct regmap *regmap)
+{
+ int status;
+
+ regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status);
+ return !(status & NAU8825_GPIO2JD1);
+}
+
+static void nau8825_restart_jack_detection(struct regmap *regmap)
+{
+ /* this will restart the entire jack detection process including MIC/GND
+ * switching and create interrupts. We have to go from 0 to 1 and back
+ * to 0 to restart.
+ */
+ regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+ NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART);
+ regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+ NAU8825_JACK_DET_RESTART, 0);
+}
+
+static void nau8825_eject_jack(struct nau8825 *nau8825)
+{
+ struct snd_soc_dapm_context *dapm = nau8825->dapm;
+ struct regmap *regmap = nau8825->regmap;
+
+ snd_soc_dapm_disable_pin(dapm, "SAR");
+ snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+ /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */
+ regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+ NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0);
+ /* ground HPL/HPR, MICGRND1/2 */
+ regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf);
+
+ snd_soc_dapm_sync(dapm);
+}
+
+static int nau8825_button_decode(int value)
+{
+ int buttons = 0;
+
+ /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */
+ if (value & BIT(0))
+ buttons |= SND_JACK_BTN_0;
+ if (value & BIT(1))
+ buttons |= SND_JACK_BTN_1;
+ if (value & BIT(2))
+ buttons |= SND_JACK_BTN_2;
+ if (value & BIT(3))
+ buttons |= SND_JACK_BTN_3;
+ if (value & BIT(4))
+ buttons |= SND_JACK_BTN_4;
+ if (value & BIT(5))
+ buttons |= SND_JACK_BTN_5;
+
+ return buttons;
+}
+
+static int nau8825_jack_insert(struct nau8825 *nau8825)
+{
+ struct regmap *regmap = nau8825->regmap;
+ struct snd_soc_dapm_context *dapm = nau8825->dapm;
+ int jack_status_reg, mic_detected;
+ int type = 0;
+
+ regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg);
+ mic_detected = (jack_status_reg >> 10) & 3;
+
+ switch (mic_detected) {
+ case 0:
+ /* no mic */
+ type = SND_JACK_HEADPHONE;
+ break;
+ case 1:
+ dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n");
+ type = SND_JACK_HEADSET;
+
+ /* Unground MICGND1 */
+ regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2,
+ 1 << 2);
+ /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */
+ regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+ NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2,
+ NAU8825_MICBIAS_JKR2);
+ /* Attach SARADC to MICGND1 */
+ regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+ NAU8825_SAR_INPUT_MASK,
+ NAU8825_SAR_INPUT_JKR2);
+
+ snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+ snd_soc_dapm_force_enable_pin(dapm, "SAR");
+ snd_soc_dapm_sync(dapm);
+ break;
+ case 2:
+ case 3:
+ dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n");
+ type = SND_JACK_HEADSET;
+
+ /* Unground MICGND2 */
+ regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2,
+ 2 << 2);
+ /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */
+ regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+ NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2,
+ NAU8825_MICBIAS_JKSLV);
+ /* Attach SARADC to MICGND2 */
+ regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+ NAU8825_SAR_INPUT_MASK,
+ NAU8825_SAR_INPUT_JKSLV);
+
+ snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+ snd_soc_dapm_force_enable_pin(dapm, "SAR");
+ snd_soc_dapm_sync(dapm);
+ break;
+ }
+
+ if (type & SND_JACK_HEADPHONE) {
+ /* Unground HPL/R */
+ regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0);
+ }
+
+ return type;
+}
+
+#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \
+ SND_JACK_BTN_2 | SND_JACK_BTN_3)
+
+static irqreturn_t nau8825_interrupt(int irq, void *data)
+{
+ struct nau8825 *nau8825 = (struct nau8825 *)data;
+ struct regmap *regmap = nau8825->regmap;
+ int active_irq, clear_irq = 0, event = 0, event_mask = 0;
+
+ regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq);
+
+ if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) ==
+ NAU8825_JACK_EJECTION_DETECTED) {
+
+ nau8825_eject_jack(nau8825);
+ event_mask |= SND_JACK_HEADSET;
+ clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK;
+ } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) {
+ int key_status;
+
+ regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS,
+ &key_status);
+
+ /* upper 8 bits of the register are for short pressed keys,
+ * lower 8 bits - for long pressed buttons
+ */
+ nau8825->button_pressed = nau8825_button_decode(
+ key_status >> 8);
+
+ event |= nau8825->button_pressed;
+ event_mask |= NAU8825_BUTTONS;
+ clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ;
+ } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) {
+ event_mask = NAU8825_BUTTONS;
+ clear_irq = NAU8825_KEY_RELEASE_IRQ;
+ } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) {
+ if (nau8825_is_jack_inserted(regmap)) {
+ event |= nau8825_jack_insert(nau8825);
+ } else {
+ dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n");
+ nau8825_eject_jack(nau8825);
+ }
+
+ event_mask |= SND_JACK_HEADSET;
+ clear_irq = NAU8825_HEADSET_COMPLETION_IRQ;
+ }
+
+ if (!clear_irq)
+ clear_irq = active_irq;
+ /* clears the rightmost interruption */
+ regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq);
+
+ if (event_mask)
+ snd_soc_jack_report(nau8825->jack, event, event_mask);
+
+ return IRQ_HANDLED;
+}
+
+static void nau8825_setup_buttons(struct nau8825 *nau8825)
+{
+ struct regmap *regmap = nau8825->regmap;
+
+ regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+ NAU8825_SAR_TRACKING_GAIN_MASK,
+ nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT);
+ regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+ NAU8825_SAR_COMPARE_TIME_MASK,
+ nau8825->sar_compare_time << NAU8825_SAR_COMPARE_TIME_SFT);
+ regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+ NAU8825_SAR_SAMPLING_TIME_MASK,
+ nau8825->sar_sampling_time << NAU8825_SAR_SAMPLING_TIME_SFT);
+
+ regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL,
+ NAU8825_KEYDET_LEVELS_NR_MASK,
+ (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT);
+ regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL,
+ NAU8825_KEYDET_HYSTERESIS_MASK,
+ nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT);
+ regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL,
+ NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK,
+ nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT);
+
+ regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1,
+ (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]);
+ regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2,
+ (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]);
+ regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3,
+ (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]);
+ regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4,
+ (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]);
+
+ /* Enable short press and release interruptions */
+ regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+ NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN,
+ 0);
+}
+
+static void nau8825_init_regs(struct nau8825 *nau8825)
+{
+ struct regmap *regmap = nau8825->regmap;
+
+ /* Enable Bias/Vmid */
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ,
+ NAU8825_BIAS_VMID, NAU8825_BIAS_VMID);
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST,
+ NAU8825_GLOBAL_BIAS_EN, NAU8825_GLOBAL_BIAS_EN);
+
+ /* VMID Tieoff */
+ regmap_update_bits(regmap, NAU8825_REG_BIAS_ADJ,
+ NAU8825_BIAS_VMID_SEL_MASK,
+ nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT);
+ /* Disable Boost Driver, Automatic Short circuit protection enable */
+ regmap_update_bits(regmap, NAU8825_REG_BOOST,
+ NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS |
+ NAU8825_SHORT_SHUTDOWN_EN,
+ NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS |
+ NAU8825_SHORT_SHUTDOWN_EN);
+
+ regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL,
+ NAU8825_JKDET_OUTPUT_EN,
+ nau8825->jkdet_enable ? 0 : NAU8825_JKDET_OUTPUT_EN);
+ regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL,
+ NAU8825_JKDET_PULL_EN,
+ nau8825->jkdet_pull_enable ? 0 : NAU8825_JKDET_PULL_EN);
+ regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL,
+ NAU8825_JKDET_PULL_UP,
+ nau8825->jkdet_pull_up ? NAU8825_JKDET_PULL_UP : 0);
+ regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+ NAU8825_JACK_POLARITY,
+ /* jkdet_polarity - 1 is for active-low */
+ nau8825->jkdet_polarity ? 0 : NAU8825_JACK_POLARITY);
+
+ regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+ NAU8825_JACK_INSERT_DEBOUNCE_MASK,
+ nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT);
+ regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+ NAU8825_JACK_EJECT_DEBOUNCE_MASK,
+ nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT);
+
+ /* Mask unneeded IRQs: 1 - disable, 0 - enable */
+ regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff);
+
+ regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+ NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage);
+
+ if (nau8825->sar_threshold_num)
+ nau8825_setup_buttons(nau8825);
+
+ /* Default oversampling/decimations settings are unusable
+ * (audible hiss). Set it to something better.
+ */
+ regmap_update_bits(regmap, NAU8825_REG_ADC_RATE,
+ NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128);
+ regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1,
+ NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128);
+}
+
+static const struct regmap_config nau8825_regmap_config = {
+ .val_bits = 16,
+ .reg_bits = 16,
+
+ .max_register = NAU8825_REG_MAX,
+ .readable_reg = nau8825_readable_reg,
+ .writeable_reg = nau8825_writeable_reg,
+ .volatile_reg = nau8825_volatile_reg,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = nau8825_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults),
+};
+
+static int nau8825_codec_probe(struct snd_soc_codec *codec)
+{
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+
+ nau8825->dapm = dapm;
+
+ /* The interrupt clock is gated by x1[10:8],
+ * one of them needs to be enabled all the time for
+ * interrupts to happen.
+ */
+ snd_soc_dapm_force_enable_pin(dapm, "DDACR");
+ snd_soc_dapm_sync(dapm);
+
+ /* Unmask interruptions. Handler uses dapm object so we can enable
+ * interruptions only after dapm is fully initialized.
+ */
+ regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0);
+ nau8825_restart_jack_detection(nau8825->regmap);
+
+ return 0;
+}
+
+/**
+ * nau8825_calc_fll_param - Calculate FLL parameters.
+ * @fll_in: external clock provided to codec.
+ * @fs: sampling rate.
+ * @fll_param: Pointer to structure of FLL parameters.
+ *
+ * Calculate FLL parameters to configure codec.
+ *
+ * Returns 0 for success or negative error code.
+ */
+static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs,
+ struct nau8825_fll *fll_param)
+{
+ u64 fvco;
+ unsigned int fref, i;
+
+ /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing
+ * freq_in by 1, 2, 4, or 8 using FLL pre-scalar.
+ * FREF = freq_in / NAU8825_FLL_REF_DIV_MASK
+ */
+ for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) {
+ fref = fll_in / fll_pre_scalar[i].param;
+ if (fref <= NAU_FREF_MAX)
+ break;
+ }
+ if (i == ARRAY_SIZE(fll_pre_scalar))
+ return -EINVAL;
+ fll_param->clk_ref_div = fll_pre_scalar[i].val;
+
+ /* Choose the FLL ratio based on FREF */
+ for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) {
+ if (fref >= fll_ratio[i].param)
+ break;
+ }
+ if (i == ARRAY_SIZE(fll_ratio))
+ return -EINVAL;
+ fll_param->ratio = fll_ratio[i].val;
+
+ /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs.
+ * FDCO must be within the 90MHz - 100MHz or the FFL cannot be
+ * guaranteed across the full range of operation.
+ * FDCO = freq_out * 2 * mclk_src_scaling
+ */
+ for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
+ fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+ if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX)
+ break;
+ }
+ if (i == ARRAY_SIZE(mclk_src_scaling))
+ return -EINVAL;
+ fll_param->mclk_src = mclk_src_scaling[i].val;
+
+ /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional
+ * input based on FDCO, FREF and FLL ratio.
+ */
+ fvco = div_u64(fvco << 16, fref * fll_param->ratio);
+ fll_param->fll_int = (fvco >> 16) & 0x3FF;
+ fll_param->fll_frac = fvco & 0xFFFF;
+ return 0;
+}
+
+static void nau8825_fll_apply(struct nau8825 *nau8825,
+ struct nau8825_fll *fll_param)
+{
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER,
+ NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src);
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1,
+ NAU8825_FLL_RATIO_MASK, fll_param->ratio);
+ /* FLL 16-bit fractional input */
+ regmap_write(nau8825->regmap, NAU8825_REG_FLL2, fll_param->fll_frac);
+ /* FLL 10-bit integer input */
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL3,
+ NAU8825_FLL_INTEGER_MASK, fll_param->fll_int);
+ /* FLL pre-scaler */
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4,
+ NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div);
+ /* select divided VCO input */
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5,
+ NAU8825_FLL_FILTER_SW_MASK, 0x0000);
+ /* FLL sigma delta modulator enable */
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6,
+ NAU8825_SDM_EN_MASK, NAU8825_SDM_EN);
+}
+
+/* freq_out must be 256*Fs in order to achieve the best performance */
+static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out)
+{
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+ struct nau8825_fll fll_param;
+ int ret, fs;
+
+ fs = freq_out / 256;
+ ret = nau8825_calc_fll_param(freq_in, fs, &fll_param);
+ if (ret < 0) {
+ dev_err(codec->dev, "Unsupported input clock %d\n", freq_in);
+ return ret;
+ }
+ dev_dbg(codec->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n",
+ fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac,
+ fll_param.fll_int, fll_param.clk_ref_div);
+
+ nau8825_fll_apply(nau8825, &fll_param);
+ mdelay(2);
+ regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER,
+ NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO);
+ return 0;
+}
+
+static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
+ unsigned int freq)
+{
+ struct regmap *regmap = nau8825->regmap;
+ int ret;
+
+ switch (clk_id) {
+ case NAU8825_CLK_MCLK:
+ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER,
+ NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK);
+ regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0);
+
+ /* We selected MCLK source but the clock itself managed externally */
+ if (!nau8825->mclk)
+ break;
+
+ if (!nau8825->mclk_freq) {
+ ret = clk_prepare_enable(nau8825->mclk);
+ if (ret) {
+ dev_err(nau8825->dev, "Unable to prepare codec mclk\n");
+ return ret;
+ }
+ }
+
+ if (nau8825->mclk_freq != freq) {
+ nau8825->mclk_freq = freq;
+
+ freq = clk_round_rate(nau8825->mclk, freq);
+ ret = clk_set_rate(nau8825->mclk, freq);
+ if (ret) {
+ dev_err(nau8825->dev, "Unable to set mclk rate\n");
+ return ret;
+ }
+ }
+
+ break;
+ case NAU8825_CLK_INTERNAL:
+ regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN,
+ NAU8825_DCO_EN);
+ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER,
+ NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO);
+
+ if (nau8825->mclk_freq) {
+ clk_disable_unprepare(nau8825->mclk);
+ nau8825->mclk_freq = 0;
+ }
+
+ break;
+ default:
+ dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id);
+ return -EINVAL;
+ }
+
+ dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq,
+ clk_id);
+ return 0;
+}
+
+static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ int source, unsigned int freq, int dir)
+{
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+
+ return nau8825_configure_sysclk(nau8825, clk_id, freq);
+}
+
+static int nau8825_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+ if (nau8825->mclk_freq) {
+ ret = clk_prepare_enable(nau8825->mclk);
+ if (ret) {
+ dev_err(nau8825->dev, "Unable to prepare codec mclk\n");
+ return ret;
+ }
+ }
+
+ ret = regcache_sync(nau8825->regmap);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+ }
+
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ if (nau8825->mclk_freq)
+ clk_disable_unprepare(nau8825->mclk);
+
+ regcache_mark_dirty(nau8825->regmap);
+ break;
+ }
+ return 0;
+}
+
+static struct snd_soc_codec_driver nau8825_codec_driver = {
+ .probe = nau8825_codec_probe,
+ .set_sysclk = nau8825_set_sysclk,
+ .set_pll = nau8825_set_pll,
+ .set_bias_level = nau8825_set_bias_level,
+ .suspend_bias_off = true,
+
+ .controls = nau8825_controls,
+ .num_controls = ARRAY_SIZE(nau8825_controls),
+ .dapm_widgets = nau8825_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets),
+ .dapm_routes = nau8825_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes),
+};
+
+static void nau8825_reset_chip(struct regmap *regmap)
+{
+ regmap_write(regmap, NAU8825_REG_RESET, 0x00);
+ regmap_write(regmap, NAU8825_REG_RESET, 0x00);
+}
+
+static void nau8825_print_device_properties(struct nau8825 *nau8825)
+{
+ int i;
+ struct device *dev = nau8825->dev;
+
+ dev_dbg(dev, "jkdet-enable: %d\n", nau8825->jkdet_enable);
+ dev_dbg(dev, "jkdet-pull-enable: %d\n", nau8825->jkdet_pull_enable);
+ dev_dbg(dev, "jkdet-pull-up: %d\n", nau8825->jkdet_pull_up);
+ dev_dbg(dev, "jkdet-polarity: %d\n", nau8825->jkdet_polarity);
+ dev_dbg(dev, "micbias-voltage: %d\n", nau8825->micbias_voltage);
+ dev_dbg(dev, "vref-impedance: %d\n", nau8825->vref_impedance);
+
+ dev_dbg(dev, "sar-threshold-num: %d\n", nau8825->sar_threshold_num);
+ for (i = 0; i < nau8825->sar_threshold_num; i++)
+ dev_dbg(dev, "sar-threshold[%d]=%d\n", i,
+ nau8825->sar_threshold[i]);
+
+ dev_dbg(dev, "sar-hysteresis: %d\n", nau8825->sar_hysteresis);
+ dev_dbg(dev, "sar-voltage: %d\n", nau8825->sar_voltage);
+ dev_dbg(dev, "sar-compare-time: %d\n", nau8825->sar_compare_time);
+ dev_dbg(dev, "sar-sampling-time: %d\n", nau8825->sar_sampling_time);
+ dev_dbg(dev, "short-key-debounce: %d\n", nau8825->key_debounce);
+ dev_dbg(dev, "jack-insert-debounce: %d\n",
+ nau8825->jack_insert_debounce);
+ dev_dbg(dev, "jack-eject-debounce: %d\n",
+ nau8825->jack_eject_debounce);
+}
+
+static int nau8825_read_device_properties(struct device *dev,
+ struct nau8825 *nau8825) {
+
+ nau8825->jkdet_enable = device_property_read_bool(dev,
+ "nuvoton,jkdet-enable");
+ nau8825->jkdet_pull_enable = device_property_read_bool(dev,
+ "nuvoton,jkdet-pull-enable");
+ nau8825->jkdet_pull_up = device_property_read_bool(dev,
+ "nuvoton,jkdet-pull-up");
+ device_property_read_u32(dev, "nuvoton,jkdet-polarity",
+ &nau8825->jkdet_polarity);
+ device_property_read_u32(dev, "nuvoton,micbias-voltage",
+ &nau8825->micbias_voltage);
+ device_property_read_u32(dev, "nuvoton,vref-impedance",
+ &nau8825->vref_impedance);
+ device_property_read_u32(dev, "nuvoton,sar-threshold-num",
+ &nau8825->sar_threshold_num);
+ device_property_read_u32_array(dev, "nuvoton,sar-threshold",
+ nau8825->sar_threshold, nau8825->sar_threshold_num);
+ device_property_read_u32(dev, "nuvoton,sar-hysteresis",
+ &nau8825->sar_hysteresis);
+ device_property_read_u32(dev, "nuvoton,sar-voltage",
+ &nau8825->sar_voltage);
+ device_property_read_u32(dev, "nuvoton,sar-compare-time",
+ &nau8825->sar_compare_time);
+ device_property_read_u32(dev, "nuvoton,sar-sampling-time",
+ &nau8825->sar_sampling_time);
+ device_property_read_u32(dev, "nuvoton,short-key-debounce",
+ &nau8825->key_debounce);
+ device_property_read_u32(dev, "nuvoton,jack-insert-debounce",
+ &nau8825->jack_insert_debounce);
+ device_property_read_u32(dev, "nuvoton,jack-eject-debounce",
+ &nau8825->jack_eject_debounce);
+
+ nau8825->mclk = devm_clk_get(dev, "mclk");
+ if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) {
+ return -EPROBE_DEFER;
+ } else if (PTR_ERR(nau8825->mclk) == -ENOENT) {
+ /* The MCLK is managed externally or not used at all */
+ nau8825->mclk = NULL;
+ dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally");
+ } else if (IS_ERR(nau8825->mclk)) {
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int nau8825_setup_irq(struct nau8825 *nau8825)
+{
+ struct regmap *regmap = nau8825->regmap;
+ int ret;
+
+ /* IRQ Output Enable */
+ regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+ NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN);
+
+ /* Enable internal VCO needed for interruptions */
+ nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0);
+
+ /* Enable DDACR needed for interrupts
+ * It is the same as force_enable_pin("DDACR") we do later
+ */
+ regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL,
+ NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR);
+
+ /* Chip needs one FSCLK cycle in order to generate interrupts,
+ * as we cannot guarantee one will be provided by the system. Turning
+ * master mode on then off enables us to generate that FSCLK cycle
+ * with a minimum of contention on the clock bus.
+ */
+ regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
+ NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER);
+ regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
+ NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE);
+
+ ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL,
+ nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+ "nau8825", nau8825);
+
+ if (ret) {
+ dev_err(nau8825->dev, "Cannot request irq %d (%d)\n",
+ nau8825->irq, ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int nau8825_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct nau8825 *nau8825 = dev_get_platdata(&i2c->dev);
+ int ret, value;
+
+ if (!nau8825) {
+ nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL);
+ if (!nau8825)
+ return -ENOMEM;
+ ret = nau8825_read_device_properties(dev, nau8825);
+ if (ret)
+ return ret;
+ }
+
+ i2c_set_clientdata(i2c, nau8825);
+
+ nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config);
+ if (IS_ERR(nau8825->regmap))
+ return PTR_ERR(nau8825->regmap);
+ nau8825->dev = dev;
+ nau8825->irq = i2c->irq;
+
+ nau8825_print_device_properties(nau8825);
+
+ nau8825_reset_chip(nau8825->regmap);
+ ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value);
+ if (ret < 0) {
+ dev_err(dev, "Failed to read device id from the NAU8825: %d\n",
+ ret);
+ return ret;
+ }
+ if ((value & NAU8825_SOFTWARE_ID_MASK) !=
+ NAU8825_SOFTWARE_ID_NAU8825) {
+ dev_err(dev, "Not a NAU8825 chip\n");
+ return -ENODEV;
+ }
+
+ nau8825_init_regs(nau8825);
+
+ if (i2c->irq)
+ nau8825_setup_irq(nau8825);
+
+ return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver,
+ &nau8825_dai, 1);
+}
+
+static int nau8825_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id nau8825_i2c_ids[] = {
+ { "nau8825", 0 },
+ { }
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id nau8825_of_ids[] = {
+ { .compatible = "nuvoton,nau8825", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, nau8825_of_ids);
+#endif
+
+#ifdef CONFIG_ACPI
+static const struct acpi_device_id nau8825_acpi_match[] = {
+ { "10508825", 0 },
+ {},
+};
+MODULE_DEVICE_TABLE(acpi, nau8825_acpi_match);
+#endif
+
+static struct i2c_driver nau8825_driver = {
+ .driver = {
+ .name = "nau8825",
+ .of_match_table = of_match_ptr(nau8825_of_ids),
+ .acpi_match_table = ACPI_PTR(nau8825_acpi_match),
+ },
+ .probe = nau8825_i2c_probe,
+ .remove = nau8825_i2c_remove,
+ .id_table = nau8825_i2c_ids,
+};
+module_i2c_driver(nau8825_driver);
+
+MODULE_DESCRIPTION("ASoC nau8825 driver");
+MODULE_AUTHOR("Anatol Pomozov <anatol@chromium.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h
new file mode 100644
index 0000000..dff8edb
--- /dev/null
+++ b/sound/soc/codecs/nau8825.h
@@ -0,0 +1,341 @@
+/*
+ * NAU8825 ALSA SoC audio driver
+ *
+ * Copyright 2015 Google Inc.
+ * Author: Anatol Pomozov <anatol.pomozov@chrominium.org>
+ *
+ * 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 __NAU8825_H__
+#define __NAU8825_H__
+
+#define NAU8825_REG_RESET 0x00
+#define NAU8825_REG_ENA_CTRL 0x01
+#define NAU8825_REG_CLK_DIVIDER 0x03
+#define NAU8825_REG_FLL1 0x04
+#define NAU8825_REG_FLL2 0x05
+#define NAU8825_REG_FLL3 0x06
+#define NAU8825_REG_FLL4 0x07
+#define NAU8825_REG_FLL5 0x08
+#define NAU8825_REG_FLL6 0x09
+#define NAU8825_REG_FLL_VCO_RSV 0x0a
+#define NAU8825_REG_HSD_CTRL 0x0c
+#define NAU8825_REG_JACK_DET_CTRL 0x0d
+#define NAU8825_REG_INTERRUPT_MASK 0x0f
+#define NAU8825_REG_IRQ_STATUS 0x10
+#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11
+#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12
+#define NAU8825_REG_SAR_CTRL 0x13
+#define NAU8825_REG_KEYDET_CTRL 0x14
+#define NAU8825_REG_VDET_THRESHOLD_1 0x15
+#define NAU8825_REG_VDET_THRESHOLD_2 0x16
+#define NAU8825_REG_VDET_THRESHOLD_3 0x17
+#define NAU8825_REG_VDET_THRESHOLD_4 0x18
+#define NAU8825_REG_GPIO34_CTRL 0x19
+#define NAU8825_REG_GPIO12_CTRL 0x1a
+#define NAU8825_REG_TDM_CTRL 0x1b
+#define NAU8825_REG_I2S_PCM_CTRL1 0x1c
+#define NAU8825_REG_I2S_PCM_CTRL2 0x1d
+#define NAU8825_REG_LEFT_TIME_SLOT 0x1e
+#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f
+#define NAU8825_REG_BIQ_CTRL 0x20
+#define NAU8825_REG_BIQ_COF1 0x21
+#define NAU8825_REG_BIQ_COF2 0x22
+#define NAU8825_REG_BIQ_COF3 0x23
+#define NAU8825_REG_BIQ_COF4 0x24
+#define NAU8825_REG_BIQ_COF5 0x25
+#define NAU8825_REG_BIQ_COF6 0x26
+#define NAU8825_REG_BIQ_COF7 0x27
+#define NAU8825_REG_BIQ_COF8 0x28
+#define NAU8825_REG_BIQ_COF9 0x29
+#define NAU8825_REG_BIQ_COF10 0x2a
+#define NAU8825_REG_ADC_RATE 0x2b
+#define NAU8825_REG_DAC_CTRL1 0x2c
+#define NAU8825_REG_DAC_CTRL2 0x2d
+#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f
+#define NAU8825_REG_ADC_DGAIN_CTRL 0x30
+#define NAU8825_REG_MUTE_CTRL 0x31
+#define NAU8825_REG_HSVOL_CTRL 0x32
+#define NAU8825_REG_DACL_CTRL 0x33
+#define NAU8825_REG_DACR_CTRL 0x34
+#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38
+#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39
+#define NAU8825_REG_ADC_DRC_SLOPES 0x3a
+#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b
+#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45
+#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46
+#define NAU8825_REG_DAC_DRC_SLOPES 0x47
+#define NAU8825_REG_DAC_DRC_ATKDCY 0x48
+#define NAU8825_REG_IMM_MODE_CTRL 0x4c
+#define NAU8825_REG_IMM_RMS_L 0x4d
+#define NAU8825_REG_IMM_RMS_R 0x4e
+#define NAU8825_REG_CLASSG_CTRL 0x50
+#define NAU8825_REG_OPT_EFUSE_CTRL 0x51
+#define NAU8825_REG_MISC_CTRL 0x55
+#define NAU8825_REG_I2C_DEVICE_ID 0x58
+#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59
+#define NAU8825_REG_BIAS_ADJ 0x66
+#define NAU8825_REG_TRIM_SETTINGS 0x68
+#define NAU8825_REG_ANALOG_CONTROL_1 0x69
+#define NAU8825_REG_ANALOG_CONTROL_2 0x6a
+#define NAU8825_REG_ANALOG_ADC_1 0x71
+#define NAU8825_REG_ANALOG_ADC_2 0x72
+#define NAU8825_REG_RDAC 0x73
+#define NAU8825_REG_MIC_BIAS 0x74
+#define NAU8825_REG_BOOST 0x76
+#define NAU8825_REG_FEPGA 0x77
+#define NAU8825_REG_POWER_UP_CONTROL 0x7f
+#define NAU8825_REG_CHARGE_PUMP 0x80
+#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81
+#define NAU8825_REG_GENERAL_STATUS 0x82
+#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS
+
+/* ENA_CTRL (0x1) */
+#define NAU8825_ENABLE_DACR_SFT 10
+#define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT)
+#define NAU8825_ENABLE_DACL_SFT 9
+#define NAU8825_ENABLE_ADC_SFT 8
+#define NAU8825_ENABLE_SAR_SFT 1
+
+/* CLK_DIVIDER (0x3) */
+#define NAU8825_CLK_SRC_SFT 15
+#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT)
+#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT)
+#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT)
+#define NAU8825_CLK_MCLK_SRC_MASK (0xf << 0)
+
+/* FLL1 (0x04) */
+#define NAU8825_FLL_RATIO_MASK (0x7f << 0)
+
+/* FLL3 (0x06) */
+#define NAU8825_FLL_INTEGER_MASK (0x3ff << 0)
+
+/* FLL4 (0x07) */
+#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10)
+
+/* FLL5 (0x08) */
+#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14)
+
+/* FLL6 (0x9) */
+#define NAU8825_DCO_EN_MASK (0x1 << 15)
+#define NAU8825_DCO_EN (0x1 << 15)
+#define NAU8825_DCO_DIS (0x0 << 15)
+#define NAU8825_SDM_EN_MASK (0x1 << 14)
+#define NAU8825_SDM_EN (0x1 << 14)
+#define NAU8825_SDM_DIS (0x0 << 14)
+
+/* HSD_CTRL (0xc) */
+#define NAU8825_HSD_AUTO_MODE (1 << 6)
+/* 0 - short to GND, 1 - open */
+#define NAU8825_SPKR_DWN1R (1 << 1)
+#define NAU8825_SPKR_DWN1L (1 << 0)
+
+/* JACK_DET_CTRL (0xd) */
+#define NAU8825_JACK_DET_RESTART (1 << 9)
+#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5
+#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT)
+#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2
+#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT)
+#define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */
+
+/* INTERRUPT_MASK (0xf) */
+#define NAU8825_IRQ_OUTPUT_EN (1 << 11)
+#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10)
+#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7)
+#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5)
+#define NAU8825_IRQ_EJECT_EN (1 << 2)
+
+/* IRQ_STATUS (0x10) */
+#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10)
+#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9)
+#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8)
+#define NAU8825_KEY_IRQ_MASK (0x7 << 5)
+#define NAU8825_KEY_RELEASE_IRQ (1 << 7)
+#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6)
+#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5)
+#define NAU8825_MIC_DETECTION_IRQ (1 << 4)
+#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2)
+#define NAU8825_JACK_EJECTION_DETECTED (1 << 2)
+#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0)
+#define NAU8825_JACK_INSERTION_DETECTED (1 << 0)
+
+/* INTERRUPT_DIS_CTRL (0x12) */
+#define NAU8825_IRQ_HEADSET_COMPLETE_DIS (1 << 10)
+#define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7)
+#define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5)
+#define NAU8825_IRQ_EJECT_DIS (1 << 2)
+
+/* SAR_CTRL (0x13) */
+#define NAU8825_SAR_ADC_EN_SFT 12
+#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT)
+#define NAU8825_SAR_INPUT_MASK (1 << 11)
+#define NAU8825_SAR_INPUT_JKSLV (1 << 11)
+#define NAU8825_SAR_INPUT_JKR2 (0 << 11)
+#define NAU8825_SAR_TRACKING_GAIN_SFT 8
+#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT)
+#define NAU8825_SAR_COMPARE_TIME_SFT 2
+#define NAU8825_SAR_COMPARE_TIME_MASK (3 << 2)
+#define NAU8825_SAR_SAMPLING_TIME_SFT 0
+#define NAU8825_SAR_SAMPLING_TIME_MASK (3 << 0)
+
+/* KEYDET_CTRL (0x14) */
+#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12
+#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT)
+#define NAU8825_KEYDET_LEVELS_NR_SFT 8
+#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8)
+#define NAU8825_KEYDET_HYSTERESIS_SFT 0
+#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf
+
+/* GPIO12_CTRL (0x1a) */
+#define NAU8825_JKDET_PULL_UP (1 << 11) /* 0 - pull down, 1 - pull up */
+#define NAU8825_JKDET_PULL_EN (1 << 9) /* 0 - enable pull, 1 - disable */
+#define NAU8825_JKDET_OUTPUT_EN (1 << 8) /* 0 - enable input, 1 - enable output */
+
+/* I2S_PCM_CTRL1 (0x1c) */
+#define NAU8825_I2S_BP_SFT 7
+#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT)
+#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT)
+#define NAU8825_I2S_PCMB_SFT 6
+#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT)
+#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT)
+#define NAU8825_I2S_DL_SFT 2
+#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DF_SFT 0
+#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT)
+
+/* I2S_PCM_CTRL2 (0x1d) */
+#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */
+#define NAU8825_I2S_MS_SFT 3
+#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT)
+#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT)
+#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT)
+
+/* ADC_RATE (0x2b) */
+#define NAU8825_ADC_SYNC_DOWN_SFT 0
+#define NAU8825_ADC_SYNC_DOWN_MASK 0x3
+#define NAU8825_ADC_SYNC_DOWN_32 0
+#define NAU8825_ADC_SYNC_DOWN_64 1
+#define NAU8825_ADC_SYNC_DOWN_128 2
+#define NAU8825_ADC_SYNC_DOWN_256 3
+
+/* DAC_CTRL1 (0x2c) */
+#define NAU8825_DAC_CLIP_OFF (1 << 7)
+#define NAU8825_DAC_OVERSAMPLE_SFT 0
+#define NAU8825_DAC_OVERSAMPLE_MASK 0x7
+#define NAU8825_DAC_OVERSAMPLE_64 0
+#define NAU8825_DAC_OVERSAMPLE_256 1
+#define NAU8825_DAC_OVERSAMPLE_128 2
+#define NAU8825_DAC_OVERSAMPLE_32 4
+
+/* MUTE_CTRL (0x31) */
+#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9)
+#define NAU8825_DAC_SOFT_MUTE (1 << 9)
+
+/* HSVOL_CTRL (0x32) */
+#define NAU8825_HP_MUTE (1 << 15)
+
+/* DACL_CTRL (0x33) */
+#define NAU8825_DACL_CH_SEL_SFT 9
+
+/* DACR_CTRL (0x34) */
+#define NAU8825_DACR_CH_SEL_SFT 9
+
+/* I2C_DEVICE_ID (0x58) */
+#define NAU8825_GPIO2JD1 (1 << 7)
+#define NAU8825_SOFTWARE_ID_MASK 0x3
+#define NAU8825_SOFTWARE_ID_NAU8825 0x0
+
+/* BIAS_ADJ (0x66) */
+#define NAU8825_BIAS_VMID (1 << 6)
+#define NAU8825_BIAS_VMID_SEL_SFT 4
+#define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT)
+
+/* ANALOG_CONTROL_2 (0x6a) */
+#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12)
+#define NAU8825_DAC_CAPACITOR_MSB (1 << 1)
+#define NAU8825_DAC_CAPACITOR_LSB (1 << 0)
+
+/* ANALOG_ADC_2 (0x72) */
+#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8)
+#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8)
+#define NAU8825_ADC_VREFSEL_VMID (1 << 8)
+#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8)
+#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8)
+#define NAU8825_POWERUP_ADCL (1 << 6)
+
+/* MIC_BIAS (0x74) */
+#define NAU8825_MICBIAS_JKSLV (1 << 14)
+#define NAU8825_MICBIAS_JKR2 (1 << 12)
+#define NAU8825_MICBIAS_POWERUP_SFT 8
+#define NAU8825_MICBIAS_VOLTAGE_SFT 0
+#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7
+
+/* BOOST (0x76) */
+#define NAU8825_PRECHARGE_DIS (1 << 13)
+#define NAU8825_GLOBAL_BIAS_EN (1 << 12)
+#define NAU8825_HP_BOOST_G_DIS (1 << 8)
+#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6)
+
+/* POWER_UP_CONTROL (0x7f) */
+#define NAU8825_POWERUP_INTEGR_R (1 << 5)
+#define NAU8825_POWERUP_INTEGR_L (1 << 4)
+#define NAU8825_POWERUP_DRV_IN_R (1 << 3)
+#define NAU8825_POWERUP_DRV_IN_L (1 << 2)
+#define NAU8825_POWERUP_HP_DRV_R (1 << 1)
+#define NAU8825_POWERUP_HP_DRV_L (1 << 0)
+
+/* CHARGE_PUMP (0x80) */
+#define NAU8825_JAMNODCLOW (1 << 10)
+#define NAU8825_POWER_DOWN_DACR (1 << 9)
+#define NAU8825_POWER_DOWN_DACL (1 << 8)
+#define NAU8825_CHANRGE_PUMP_EN (1 << 5)
+
+
+/* System Clock Source */
+enum {
+ NAU8825_CLK_MCLK = 0,
+ NAU8825_CLK_INTERNAL,
+};
+
+struct nau8825 {
+ struct device *dev;
+ struct regmap *regmap;
+ struct snd_soc_dapm_context *dapm;
+ struct snd_soc_jack *jack;
+ struct clk *mclk;
+ int irq;
+ int mclk_freq; /* 0 - mclk is disabled */
+ int button_pressed;
+ int micbias_voltage;
+ int vref_impedance;
+ bool jkdet_enable;
+ bool jkdet_pull_enable;
+ bool jkdet_pull_up;
+ int jkdet_polarity;
+ int sar_threshold_num;
+ int sar_threshold[8];
+ int sar_hysteresis;
+ int sar_voltage;
+ int sar_compare_time;
+ int sar_sampling_time;
+ int key_debounce;
+ int jack_insert_debounce;
+ int jack_eject_debounce;
+};
+
+int nau8825_enable_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *jack);
+
+
+#endif /* __NAU8825_H__ */
diff --git a/sound/soc/codecs/rl6347a.c b/sound/soc/codecs/rl6347a.c
index 91d5166..a4b910e 100644
--- a/sound/soc/codecs/rl6347a.c
+++ b/sound/soc/codecs/rl6347a.c
@@ -11,25 +11,8 @@
*/
#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/delay.h>
-#include <linux/pm.h>
#include <linux/i2c.h>
-#include <linux/platform_device.h>
-#include <linux/spi/spi.h>
-#include <linux/dmi.h>
-#include <linux/acpi.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <sound/initval.h>
-#include <sound/tlv.h>
-#include <sound/jack.h>
-#include <linux/workqueue.h>
-#include <sound/hda_verbs.h>
+#include <linux/regmap.h>
#include "rl6347a.h"
diff --git a/sound/soc/codecs/rl6347a.h b/sound/soc/codecs/rl6347a.h
index 1cb56e5..e127919 100644
--- a/sound/soc/codecs/rl6347a.h
+++ b/sound/soc/codecs/rl6347a.h
@@ -12,6 +12,8 @@
#ifndef __RL6347A_H__
#define __RL6347A_H__
+#include <sound/hda_verbs.h>
+
#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
#define RL6347A_VENDOR_REGISTERS 0x20
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index bd93658..af2ed77 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -29,7 +29,6 @@
#include <sound/jack.h>
#include <linux/workqueue.h>
#include <sound/rt286.h>
-#include <sound/hda_verbs.h>
#include "rl6347a.h"
#include "rt286.h"
@@ -38,7 +37,7 @@
#define RT288_VENDOR_ID 0x10ec0288
struct rt286_priv {
- const struct reg_default *index_cache;
+ struct reg_default *index_cache;
int index_cache_size;
struct regmap *regmap;
struct snd_soc_codec *codec;
@@ -1161,7 +1160,11 @@
return -ENODEV;
}
- rt286->index_cache = rt286_index_def;
+ rt286->index_cache = devm_kmemdup(&i2c->dev, rt286_index_def,
+ sizeof(rt286_index_def), GFP_KERNEL);
+ if (!rt286->index_cache)
+ return -ENOMEM;
+
rt286->index_cache_size = INDEX_CACHE_SIZE;
rt286->i2c = i2c;
i2c_set_clientdata(i2c, rt286);
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index f823eb5..b3f795c 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -28,7 +28,6 @@
#include <sound/jack.h>
#include <linux/workqueue.h>
#include <sound/rt298.h>
-#include <sound/hda_verbs.h>
#include "rl6347a.h"
#include "rt298.h"
@@ -49,7 +48,7 @@
int is_hp_in;
};
-static struct reg_default rt298_index_def[] = {
+static const struct reg_default rt298_index_def[] = {
{ 0x01, 0xa5a8 },
{ 0x02, 0x8e95 },
{ 0x03, 0x0002 },
@@ -129,7 +128,7 @@
case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0):
return true;
default:
- return true;
+ return false;
}
@@ -1165,7 +1164,11 @@
return -ENODEV;
}
- rt298->index_cache = rt298_index_def;
+ rt298->index_cache = devm_kmemdup(&i2c->dev, rt298_index_def,
+ sizeof(rt298_index_def), GFP_KERNEL);
+ if (!rt298->index_cache)
+ return -ENOMEM;
+
rt298->index_cache_size = INDEX_CACHE_SIZE;
rt298->i2c = i2c;
i2c_set_clientdata(i2c, rt298);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index e1ceeb8..f2beb1a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -405,11 +405,14 @@
SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL,
RT5640_L_VOL_SFT, RT5640_R_VOL_SFT,
175, 0, dac_vol_tlv),
- /* IN1/IN2 Control */
+ /* IN1/IN2/IN3 Control */
SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2,
RT5640_BST_SFT1, 8, 0, bst_tlv),
SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4,
RT5640_BST_SFT2, 8, 0, bst_tlv),
+ SOC_SINGLE_TLV("IN3 Boost", RT5640_IN1_IN2,
+ RT5640_BST_SFT2, 8, 0, bst_tlv),
+
/* INL/INR Volume Control */
SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL,
RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT,
@@ -598,6 +601,8 @@
RT5640_M_HP_L_RM_L_SFT, 1, 1),
SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER,
RT5640_M_IN_L_RM_L_SFT, 1, 1),
+ SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_L2_MIXER,
+ RT5640_M_BST2_RM_L_SFT, 1, 1),
SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER,
RT5640_M_BST4_RM_L_SFT, 1, 1),
SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER,
@@ -611,6 +616,8 @@
RT5640_M_HP_R_RM_R_SFT, 1, 1),
SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER,
RT5640_M_IN_R_RM_R_SFT, 1, 1),
+ SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_R2_MIXER,
+ RT5640_M_BST2_RM_R_SFT, 1, 1),
SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER,
RT5640_M_BST4_RM_R_SFT, 1, 1),
SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER,
@@ -1065,6 +1072,8 @@
SND_SOC_DAPM_INPUT("IN1N"),
SND_SOC_DAPM_INPUT("IN2P"),
SND_SOC_DAPM_INPUT("IN2N"),
+ SND_SOC_DAPM_INPUT("IN3P"),
+ SND_SOC_DAPM_INPUT("IN3N"),
SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -1081,6 +1090,8 @@
RT5640_PWR_BST1_BIT, 0, NULL, 0),
SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2,
RT5640_PWR_BST4_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("BST3", RT5640_PWR_ANLG2,
+ RT5640_PWR_BST2_BIT, 0, NULL, 0),
/* Input Volume */
SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL,
RT5640_PWR_IN_L_BIT, 0, NULL, 0),
@@ -1310,6 +1321,7 @@
static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
{"IN1P", NULL, "LDO2"},
{"IN2P", NULL, "LDO2"},
+ {"IN3P", NULL, "LDO2"},
{"DMIC L1", NULL, "DMIC1"},
{"DMIC R1", NULL, "DMIC1"},
@@ -1320,18 +1332,22 @@
{"BST1", NULL, "IN1N"},
{"BST2", NULL, "IN2P"},
{"BST2", NULL, "IN2N"},
+ {"BST3", NULL, "IN3P"},
+ {"BST3", NULL, "IN3N"},
{"INL VOL", NULL, "IN2P"},
{"INR VOL", NULL, "IN2N"},
{"RECMIXL", "HPOL Switch", "HPOL"},
{"RECMIXL", "INL Switch", "INL VOL"},
+ {"RECMIXL", "BST3 Switch", "BST3"},
{"RECMIXL", "BST2 Switch", "BST2"},
{"RECMIXL", "BST1 Switch", "BST1"},
{"RECMIXL", "OUT MIXL Switch", "OUT MIXL"},
{"RECMIXR", "HPOR Switch", "HPOR"},
{"RECMIXR", "INR Switch", "INR VOL"},
+ {"RECMIXR", "BST3 Switch", "BST3"},
{"RECMIXR", "BST2 Switch", "BST2"},
{"RECMIXR", "BST1 Switch", "BST1"},
{"RECMIXR", "OUT MIXR Switch", "OUT MIXR"},
@@ -2260,6 +2276,10 @@
regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
RT5640_IN_DF2, RT5640_IN_DF2);
+ if (rt5640->pdata.in3_diff)
+ regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2,
+ RT5640_IN_DF2, RT5640_IN_DF2);
+
rt5640->hp_mute = 1;
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 5c101af..2813237 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -42,6 +42,8 @@
#define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING))
+#define RT5645_HWEQ_NUM 57
+
static const struct regmap_range_cfg rt5645_ranges[] = {
{
.name = "PR",
@@ -224,6 +226,11 @@
{ 0xff, 0x6308 },
};
+struct rt5645_eq_param_s {
+ unsigned short reg;
+ unsigned short val;
+};
+
static const char *const rt5645_supply_names[] = {
"avdd",
"cpvdd",
@@ -240,6 +247,7 @@
struct snd_soc_jack *btn_jack;
struct delayed_work jack_detect_work;
struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)];
+ struct rt5645_eq_param_s *eq_param;
int codec_type;
int sysclk;
@@ -469,6 +477,94 @@
8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0)
);
+/* {-6, -4.5, -3, -1.5, 0, 0.82, 1.58, 2.28} dB */
+static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv,
+ 0, 4, TLV_DB_SCALE_ITEM(-600, 150, 0),
+ 5, 5, TLV_DB_SCALE_ITEM(82, 0, 0),
+ 6, 6, TLV_DB_SCALE_ITEM(158, 0, 0),
+ 7, 7, TLV_DB_SCALE_ITEM(228, 0, 0)
+);
+
+static int rt5645_hweq_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s);
+
+ return 0;
+}
+
+static int rt5645_hweq_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+ struct rt5645_eq_param_s *eq_param =
+ (struct rt5645_eq_param_s *)ucontrol->value.bytes.data;
+ int i;
+
+ for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+ eq_param[i].reg = cpu_to_be16(rt5645->eq_param[i].reg);
+ eq_param[i].val = cpu_to_be16(rt5645->eq_param[i].val);
+ }
+
+ return 0;
+}
+
+static bool rt5645_validate_hweq(unsigned short reg)
+{
+ if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) |
+ (reg == RT5645_EQ_CTRL2))
+ return true;
+
+ return false;
+}
+
+static int rt5645_hweq_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+ struct rt5645_eq_param_s *eq_param =
+ (struct rt5645_eq_param_s *)ucontrol->value.bytes.data;
+ int i;
+
+ for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+ eq_param[i].reg = be16_to_cpu(eq_param[i].reg);
+ eq_param[i].val = be16_to_cpu(eq_param[i].val);
+ }
+
+ /* The final setting of the table should be RT5645_EQ_CTRL2 */
+ for (i = RT5645_HWEQ_NUM - 1; i >= 0; i--) {
+ if (eq_param[i].reg == 0)
+ continue;
+ else if (eq_param[i].reg != RT5645_EQ_CTRL2)
+ return 0;
+ else
+ break;
+ }
+
+ for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+ if (!rt5645_validate_hweq(eq_param[i].reg) &&
+ eq_param[i].reg != 0)
+ return 0;
+ else if (eq_param[i].reg == 0)
+ break;
+ }
+
+ memcpy(rt5645->eq_param, eq_param,
+ RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s));
+
+ return 0;
+}
+
+#define RT5645_HWEQ(xname) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = rt5645_hweq_info, \
+ .get = rt5645_hweq_get, \
+ .put = rt5645_hweq_put \
+}
+
static const struct snd_kcontrol_new rt5645_snd_controls[] = {
/* Speaker Output Volume */
SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL,
@@ -476,6 +572,10 @@
SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv),
+ /* ClassD modulator Speaker Gain Ratio */
+ SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO,
+ RT5645_SPK_G_CLSD_SFT, 7, 0, spk_clsd_tlv),
+
/* Headphone Output Volume */
SOC_DOUBLE("Headphone Channel Switch", RT5645_HP_VOL,
RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1),
@@ -529,6 +629,7 @@
/* I2S2 function select */
SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT,
1, 1),
+ RT5645_HWEQ("Speaker HWEQ"),
};
/**
@@ -619,6 +720,22 @@
}
+static int rt5645_enable_hweq(struct snd_soc_codec *codec)
+{
+ struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+ if (rt5645_validate_hweq(rt5645->eq_param[i].reg))
+ regmap_write(rt5645->regmap, rt5645->eq_param[i].reg,
+ rt5645->eq_param[i].val);
+ else
+ break;
+ }
+
+ return 0;
+}
+
/**
* rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters
* @codec: SoC audio codec device.
@@ -1523,6 +1640,7 @@
switch (event) {
case SND_SOC_DAPM_POST_PMU:
+ rt5645_enable_hweq(codec);
snd_soc_update_bits(codec, RT5645_PWR_DIG1,
RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R |
RT5645_PWR_CLS_D_L,
@@ -1531,6 +1649,7 @@
break;
case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_write(codec, RT5645_EQ_CTRL2, 0);
snd_soc_update_bits(codec, RT5645_PWR_DIG1,
RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R |
RT5645_PWR_CLS_D_L, 0);
@@ -2733,6 +2852,10 @@
snd_soc_update_bits(codec, RT5645_PWR_ANLG1,
RT5645_PWR_FV1 | RT5645_PWR_FV2,
RT5645_PWR_FV1 | RT5645_PWR_FV2);
+ if (rt5645->en_button_func &&
+ snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
+ queue_delayed_work(system_power_efficient_wq,
+ &rt5645->jack_detect_work, msecs_to_jiffies(0));
break;
case SND_SOC_BIAS_OFF:
@@ -2829,6 +2952,9 @@
snd_soc_dapm_sync(dapm);
rt5645->jack_type = SND_JACK_HEADPHONE;
}
+ if (rt5645->pdata.jd_invert)
+ regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+ RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
} else { /* jack out */
rt5645->jack_type = 0;
@@ -2847,6 +2973,9 @@
snd_soc_dapm_disable_pin(dapm, "LDO2");
snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
snd_soc_dapm_sync(dapm);
+ if (rt5645->pdata.jd_invert)
+ regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+ RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR);
}
return rt5645->jack_type;
@@ -3038,6 +3167,9 @@
snd_soc_dapm_sync(dapm);
}
+ rt5645->eq_param = devm_kzalloc(codec->dev,
+ RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), GFP_KERNEL);
+
return 0;
}
@@ -3098,7 +3230,7 @@
.capture = {
.stream_name = "AIF1 Capture",
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 4,
.rates = RT5645_STEREO_RATES,
.formats = RT5645_FORMATS,
},
@@ -3209,9 +3341,42 @@
DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
},
},
+ {
+ .ident = "Google Reks",
+ .callback = strago_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Reks"),
+ },
+ },
{ }
};
+static struct rt5645_platform_data buddy_platform_data = {
+ .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
+ .dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
+ .jd_mode = 3,
+ .jd_invert = true,
+};
+
+static int buddy_quirk_cb(const struct dmi_system_id *id)
+{
+ rt5645_pdata = &buddy_platform_data;
+
+ return 1;
+}
+
+static struct dmi_system_id dmi_platform_intel_broadwell[] = {
+ {
+ .ident = "Chrome Buddy",
+ .callback = buddy_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"),
+ },
+ },
+ { }
+};
+
+
static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev)
{
rt5645->pdata.in2_diff = device_property_read_bool(dev,
@@ -3244,7 +3409,8 @@
if (pdata)
rt5645->pdata = *pdata;
- else if (dmi_check_system(dmi_platform_intel_braswell))
+ else if (dmi_check_system(dmi_platform_intel_braswell) ||
+ dmi_check_system(dmi_platform_intel_broadwell))
rt5645->pdata = *rt5645_pdata;
else
rt5645_parse_dt(rt5645, &i2c->dev);
@@ -3472,6 +3638,8 @@
RT5645_CBJ_MN_JD);
regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN,
0);
+ msleep(20);
+ regmap_write(rt5645->regmap, RT5645_RESET, 0);
}
static struct i2c_driver rt5645_i2c_driver = {
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 8c964cfb..093e46d 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -621,14 +621,14 @@
#define RT5645_G_OM_L_SM_L_SFT 6
#define RT5645_M_BST1_L_SM_L (0x1 << 5)
#define RT5645_M_BST1_L_SM_L_SFT 5
-#define RT5645_M_IN_L_SM_L (0x1 << 3)
-#define RT5645_M_IN_L_SM_L_SFT 3
-#define RT5645_M_DAC_L1_SM_L (0x1 << 1)
-#define RT5645_M_DAC_L1_SM_L_SFT 1
-#define RT5645_M_DAC_L2_SM_L (0x1 << 2)
-#define RT5645_M_DAC_L2_SM_L_SFT 2
#define RT5645_M_BST3_L_SM_L (0x1 << 4)
#define RT5645_M_BST3_L_SM_L_SFT 4
+#define RT5645_M_IN_L_SM_L (0x1 << 3)
+#define RT5645_M_IN_L_SM_L_SFT 3
+#define RT5645_M_DAC_L2_SM_L (0x1 << 2)
+#define RT5645_M_DAC_L2_SM_L_SFT 2
+#define RT5645_M_DAC_L1_SM_L (0x1 << 1)
+#define RT5645_M_DAC_L1_SM_L_SFT 1
/* SPK Right Mixer Control (0x47) */
#define RT5645_G_RM_R_SM_R_MASK (0x3 << 14)
@@ -643,14 +643,14 @@
#define RT5645_G_OM_R_SM_R_SFT 6
#define RT5645_M_BST2_R_SM_R (0x1 << 5)
#define RT5645_M_BST2_R_SM_R_SFT 5
-#define RT5645_M_IN_R_SM_R (0x1 << 3)
-#define RT5645_M_IN_R_SM_R_SFT 3
-#define RT5645_M_DAC_R1_SM_R (0x1 << 1)
-#define RT5645_M_DAC_R1_SM_R_SFT 1
-#define RT5645_M_DAC_R2_SM_R (0x1 << 2)
-#define RT5645_M_DAC_R2_SM_R_SFT 2
#define RT5645_M_BST3_R_SM_R (0x1 << 4)
#define RT5645_M_BST3_R_SM_R_SFT 4
+#define RT5645_M_IN_R_SM_R (0x1 << 3)
+#define RT5645_M_IN_R_SM_R_SFT 3
+#define RT5645_M_DAC_R2_SM_R (0x1 << 2)
+#define RT5645_M_DAC_R2_SM_R_SFT 2
+#define RT5645_M_DAC_R1_SM_R (0x1 << 1)
+#define RT5645_M_DAC_R1_SM_R_SFT 1
/* SPOLMIX Control (0x48) */
#define RT5645_M_DAC_L1_SPM_L (0x1 << 15)
@@ -670,13 +670,17 @@
#define RT5645_M_SV_R_SPM_R (0x1 << 0)
#define RT5645_M_SV_R_SPM_R_SFT 0
+/* SPOMIX Ratio Control (0x4a) */
+#define RT5645_SPK_G_CLSD_MASK (0x7 << 0)
+#define RT5645_SPK_G_CLSD_SFT 0
+
/* Mono Output Mixer Control (0x4c) */
+#define RT5645_G_MONOMIX_MASK (0x1 << 10)
+#define RT5645_G_MONOMIX_SFT 10
#define RT5645_M_OV_L_MM (0x1 << 9)
#define RT5645_M_OV_L_MM_SFT 9
#define RT5645_M_DAC_L2_MA (0x1 << 8)
#define RT5645_M_DAC_L2_MA_SFT 8
-#define RT5645_G_MONOMIX_MASK (0x1 << 10)
-#define RT5645_G_MONOMIX_SFT 10
#define RT5645_M_BST2_MM (0x1 << 4)
#define RT5645_M_BST2_MM_SFT 4
#define RT5645_M_DAC_R1_MM (0x1 << 3)
@@ -779,8 +783,6 @@
#define RT5645_PWR_CLS_D_R_BIT 9
#define RT5645_PWR_CLS_D_L (0x1 << 8)
#define RT5645_PWR_CLS_D_L_BIT 8
-#define RT5645_PWR_ADC_R (0x1 << 1)
-#define RT5645_PWR_ADC_R_BIT 1
#define RT5645_PWR_DAC_L2 (0x1 << 7)
#define RT5645_PWR_DAC_L2_BIT 7
#define RT5645_PWR_DAC_R2 (0x1 << 6)
@@ -1628,6 +1630,10 @@
#define RT5645_OT_P_NOR (0x0 << 10)
#define RT5645_OT_P_INV (0x1 << 10)
#define RT5645_IRQ_JD_1_1_EN (0x1 << 9)
+#define RT5645_JD_1_1_MASK (0x1 << 7)
+#define RT5645_JD_1_1_SFT 7
+#define RT5645_JD_1_1_NOR (0x0 << 7)
+#define RT5645_JD_1_1_INV (0x1 << 7)
/* IRQ Control 2 (0xbe) */
#define RT5645_IRQ_MB1_OC_MASK (0x1 << 15)
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index ddb0203..86b81a6 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -723,17 +723,11 @@
.num_dapm_routes = ARRAY_SIZE(ssm2518_routes),
};
-static bool ssm2518_register_volatile(struct device *dev, unsigned int reg)
-{
- return false;
-}
-
static const struct regmap_config ssm2518_regmap_config = {
.val_bits = 8,
.reg_bits = 8,
.max_register = SSM2518_REG_DRC_9,
- .volatile_reg = ssm2518_register_volatile,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = ssm2518_reg_defaults,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8739126..a564759 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -80,6 +80,7 @@
unsigned int sysclk;
unsigned int dai_fmt;
unsigned int tdm_delay;
+ unsigned int slot_width;
struct list_head list;
int master;
int gpio_reset;
@@ -1025,10 +1026,14 @@
u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
u16 d, pll_d = 1;
int clk;
+ int width = aic3x->slot_width;
+
+ if (!width)
+ width = params_width(params);
/* select data word length */
data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
- switch (params_width(params)) {
+ switch (width) {
case 16:
break;
case 20:
@@ -1170,12 +1175,16 @@
struct snd_soc_codec *codec = dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int delay = 0;
+ int width = aic3x->slot_width;
+
+ if (!width)
+ width = substream->runtime->sample_bits;
/* TDM slot selection only valid in DSP_A/_B mode */
if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
- delay += (aic3x->tdm_delay + 1);
+ delay += (aic3x->tdm_delay*width + 1);
else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
- delay += aic3x->tdm_delay;
+ delay += aic3x->tdm_delay*width;
/* Configure data delay */
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
@@ -1296,7 +1305,20 @@
return -EINVAL;
}
- aic3x->tdm_delay = lsb * slot_width;
+ switch (slot_width) {
+ case 16:
+ case 20:
+ case 24:
+ case 32:
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported slot width %d\n", slot_width);
+ return -EINVAL;
+ }
+
+
+ aic3x->tdm_delay = lsb;
+ aic3x->slot_width = slot_width;
/* DOUT in high-impedance on inactive bit clocks */
snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 2713e18..a5a4e9f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1612,19 +1612,16 @@
return;
/* Set the constraints according to the already configured stream */
- snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ snd_pcm_hw_constraint_single(slv_substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
- twl4030->rate,
twl4030->rate);
- snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ snd_pcm_hw_constraint_single(slv_substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- twl4030->sample_bits,
twl4030->sample_bits);
- snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ snd_pcm_hw_constraint_single(slv_substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
- twl4030->channels,
twl4030->channels);
}
@@ -1669,9 +1666,9 @@
/* In option2 4 channel is not supported, set the
* constraint for the first stream for channels, the
* second stream will 'inherit' this cosntraint */
- snd_pcm_hw_constraint_minmax(substream->runtime,
+ snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
- 2, 2);
+ 2);
}
twl4030->master_substream = substream;
}
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index e190263..e4c694c 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -150,14 +150,12 @@
master_runtime->sample_bits,
master_runtime->rate);
- snd_pcm_hw_constraint_minmax(substream->runtime,
+ snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
- master_runtime->rate,
master_runtime->rate);
- snd_pcm_hw_constraint_minmax(substream->runtime,
+ snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- master_runtime->sample_bits,
master_runtime->sample_bits);
uda134x->slave_substream = substream;
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 80fb1dc..7693c11 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -307,11 +307,10 @@
switch (wl1273->mode) {
case WL1273_MODE_BT:
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- 8000, 8000);
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1);
+ snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, 8000);
+ snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 1);
break;
case WL1273_MODE_FM_RX:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 786abd0..a67ea10 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -620,7 +620,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
- int anc_active = ucontrol->value.integer.value[0];
+ unsigned int anc_active = ucontrol->value.integer.value[0];
int ret;
if (anc_active > 1)
@@ -653,7 +653,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
- int val = ucontrol->value.integer.value[0];
+ unsigned int val = ucontrol->value.integer.value[0];
int ret;
if (val > 1)
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 9756578..c04c0bc 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -38,6 +38,12 @@
struct wm5110_priv {
struct arizona_priv core;
struct arizona_fll fll[2];
+
+ unsigned int in_value;
+ int in_pre_pending;
+ int in_post_pending;
+
+ unsigned int in_pga_cache[6];
};
static const struct wm_adsp_region wm5110_dsp1_regions[] = {
@@ -428,6 +434,127 @@
return ret;
}
+static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct snd_soc_card *card = dapm->card;
+ int ret;
+
+ /*
+ * PGA Volume is also used as part of the enable sequence, so
+ * usage of it should be avoided whilst that is running.
+ */
+ mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_get_volsw_range(kcontrol, ucontrol);
+
+ mutex_unlock(&card->dapm_mutex);
+
+ return ret;
+}
+
+static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct snd_soc_card *card = dapm->card;
+ int ret;
+
+ /*
+ * PGA Volume is also used as part of the enable sequence, so
+ * usage of it should be avoided whilst that is running.
+ */
+ mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_put_volsw_range(kcontrol, ucontrol);
+
+ mutex_unlock(&card->dapm_mutex);
+
+ return ret;
+}
+
+static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ unsigned int reg, mask;
+ struct reg_sequence analog_seq[] = {
+ { 0x80, 0x3 },
+ { 0x35d, 0 },
+ { 0x80, 0x0 },
+ };
+
+ reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4);
+ mask = ARIZONA_IN1L_PGA_VOL_MASK;
+
+ switch (event) {
+ case SND_SOC_DAPM_WILL_PMU:
+ wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2);
+ wm5110->in_pre_pending++;
+ wm5110->in_post_pending++;
+ return 0;
+ case SND_SOC_DAPM_PRE_PMU:
+ wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg);
+
+ snd_soc_update_bits(codec, reg, mask,
+ 0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT);
+
+ wm5110->in_pre_pending--;
+ if (wm5110->in_pre_pending == 0) {
+ analog_seq[1].def = wm5110->in_value;
+ regmap_multi_reg_write_bypassed(arizona->regmap,
+ analog_seq,
+ ARRAY_SIZE(analog_seq));
+
+ msleep(55);
+
+ wm5110->in_value = 0;
+ }
+
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, reg, mask,
+ wm5110->in_pga_cache[w->shift]);
+
+ wm5110->in_post_pending--;
+ if (wm5110->in_post_pending == 0)
+ regmap_multi_reg_write_bypassed(arizona->regmap,
+ analog_seq,
+ ARRAY_SIZE(analog_seq));
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int wm5110_in_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+
+ switch (arizona->rev) {
+ case 0 ... 4:
+ if (arizona_input_analog(codec, w->shift))
+ wm5110_in_analog_ev(w, kcontrol, event);
+
+ break;
+ default:
+ break;
+ }
+
+ return arizona_in_ev(w, kcontrol, event);
+}
+
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
@@ -454,18 +581,24 @@
SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]),
SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]),
-SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
- ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
- ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
- ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
- ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
- ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
- ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
+ ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
+ ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
+ ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+ wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
@@ -896,29 +1029,35 @@
SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
- 0, NULL, 0, arizona_in_ev,
+ 0, NULL, 0, wm5110_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_WILL_PMU),
SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 15bd547..4bcf5f8 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -132,7 +132,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
- int deemph = ucontrol->value.integer.value[0];
+ unsigned int deemph = ucontrol->value.integer.value[0];
int ret = 0;
if (deemph > 1)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b011253..e4cc41e 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -452,7 +452,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
- int deemph = ucontrol->value.integer.value[0];
+ unsigned int deemph = ucontrol->value.integer.value[0];
int ret = 0;
if (deemph > 1)
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index b783743..2aa23f1 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -534,7 +534,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
- int deemph = ucontrol->value.integer.value[0];
+ unsigned int deemph = ucontrol->value.integer.value[0];
if (deemph > 1)
return -EINVAL;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 12e4435..9db00d5 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -402,7 +402,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
- int deemph = ucontrol->value.integer.value[0];
+ unsigned int deemph = ucontrol->value.integer.value[0];
if (deemph > 1)
return -EINVAL;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index dbd8840..0563753 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -201,7 +201,7 @@
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
- int deemph = ucontrol->value.integer.value[0];
+ unsigned int deemph = ucontrol->value.integer.value[0];
if (deemph > 1)
return -EINVAL;
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
new file mode 100644
index 0000000..8782dfb
--- /dev/null
+++ b/sound/soc/codecs/wm8998.c
@@ -0,0 +1,1430 @@
+/*
+ * wm8998.c -- ALSA SoC Audio driver for WM8998 codecs
+ *
+ * Copyright 2015 Cirrus Logic, Inc.
+ *
+ * Author: Richard Fitzgerald <rf@opensource.wolfsonmicro.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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm8998.h"
+
+struct wm8998_priv {
+ struct arizona_priv core;
+ struct arizona_fll fll[2];
+};
+
+static int wm8998_asrc_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ unsigned int val;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ val = snd_soc_read(codec, ARIZONA_ASRC_RATE1);
+ val &= ARIZONA_ASRC_RATE1_MASK;
+ val >>= ARIZONA_ASRC_RATE1_SHIFT;
+
+ switch (val) {
+ case 0:
+ case 1:
+ case 2:
+ val = snd_soc_read(codec,
+ ARIZONA_SAMPLE_RATE_1 + val);
+ if (val >= 0x11) {
+ dev_warn(codec->dev,
+ "Unsupported ASRC rate1 (%s)\n",
+ arizona_sample_rate_val_to_name(val));
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(codec->dev,
+ "Illegal ASRC rate1 selector (0x%x)\n",
+ val);
+ return -EINVAL;
+ }
+
+ val = snd_soc_read(codec, ARIZONA_ASRC_RATE2);
+ val &= ARIZONA_ASRC_RATE2_MASK;
+ val >>= ARIZONA_ASRC_RATE2_SHIFT;
+
+ switch (val) {
+ case 8:
+ case 9:
+ val -= 0x8;
+ val = snd_soc_read(codec,
+ ARIZONA_ASYNC_SAMPLE_RATE_1 + val);
+ if (val >= 0x11) {
+ dev_warn(codec->dev,
+ "Unsupported ASRC rate2 (%s)\n",
+ arizona_sample_rate_val_to_name(val));
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(codec->dev,
+ "Illegal ASRC rate2 selector (0x%x)\n",
+ val);
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8998_in1mux_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = wm8998->core.arizona;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int mux, inmode;
+ unsigned int mode_val, src_val;
+
+ mux = ucontrol->value.enumerated.item[0];
+ if (mux > 1)
+ return -EINVAL;
+
+ /* L and R registers have same shift and mask */
+ inmode = arizona->pdata.inmode[2 * mux];
+ src_val = mux << ARIZONA_IN1L_SRC_SHIFT;
+ if (inmode & ARIZONA_INMODE_SE)
+ src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT;
+
+ switch (arizona->pdata.inmode[0]) {
+ case ARIZONA_INMODE_DMIC:
+ if (mux)
+ mode_val = 0; /* B always analogue */
+ else
+ mode_val = 1 << ARIZONA_IN1_MODE_SHIFT;
+
+ snd_soc_update_bits(codec, ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1_MODE_MASK, mode_val);
+
+ /* IN1A is digital so L and R must change together */
+ /* src_val setting same for both registers */
+ snd_soc_update_bits(codec,
+ ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_IN1L_SRC_MASK |
+ ARIZONA_IN1L_SRC_SE_MASK, src_val);
+ snd_soc_update_bits(codec,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R,
+ ARIZONA_IN1R_SRC_MASK |
+ ARIZONA_IN1R_SRC_SE_MASK, src_val);
+ break;
+ default:
+ /* both analogue */
+ snd_soc_update_bits(codec,
+ e->reg,
+ ARIZONA_IN1L_SRC_MASK |
+ ARIZONA_IN1L_SRC_SE_MASK,
+ src_val);
+ break;
+ }
+
+ return snd_soc_dapm_mux_update_power(dapm, kcontrol,
+ ucontrol->value.enumerated.item[0],
+ e, NULL);
+}
+
+static int wm8998_in2mux_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = wm8998->core.arizona;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int mux, inmode, src_val, mode_val;
+
+ mux = ucontrol->value.enumerated.item[0];
+ if (mux > 1)
+ return -EINVAL;
+
+ inmode = arizona->pdata.inmode[1 + (2 * mux)];
+ if (inmode & ARIZONA_INMODE_DMIC)
+ mode_val = 1 << ARIZONA_IN2_MODE_SHIFT;
+ else
+ mode_val = 0;
+
+ src_val = mux << ARIZONA_IN2L_SRC_SHIFT;
+ if (inmode & ARIZONA_INMODE_SE)
+ src_val |= 1 << ARIZONA_IN2L_SRC_SE_SHIFT;
+
+ snd_soc_update_bits(codec, ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2_MODE_MASK, mode_val);
+
+ snd_soc_update_bits(codec, ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_IN2L_SRC_MASK | ARIZONA_IN2L_SRC_SE_MASK,
+ src_val);
+
+ return snd_soc_dapm_mux_update_power(dapm, kcontrol,
+ ucontrol->value.enumerated.item[0],
+ e, NULL);
+}
+
+static const char * const wm8998_inmux_texts[] = {
+ "A",
+ "B",
+};
+
+static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxl_enum,
+ ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_IN1L_SRC_SHIFT,
+ wm8998_inmux_texts);
+
+static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxr_enum,
+ ARIZONA_ADC_DIGITAL_VOLUME_1R,
+ ARIZONA_IN1R_SRC_SHIFT,
+ wm8998_inmux_texts);
+
+static const SOC_ENUM_SINGLE_DECL(wm8998_in2mux_enum,
+ ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_IN2L_SRC_SHIFT,
+ wm8998_inmux_texts);
+
+static const struct snd_kcontrol_new wm8998_in1mux[2] = {
+ SOC_DAPM_ENUM_EXT("IN1L Mux", wm8998_in1muxl_enum,
+ snd_soc_dapm_get_enum_double, wm8998_in1mux_put),
+ SOC_DAPM_ENUM_EXT("IN1R Mux", wm8998_in1muxr_enum,
+ snd_soc_dapm_get_enum_double, wm8998_in1mux_put),
+};
+
+static const struct snd_kcontrol_new wm8998_in2mux =
+ SOC_DAPM_ENUM_EXT("IN2 Mux", wm8998_in2mux_enum,
+ snd_soc_dapm_get_enum_double, wm8998_in2mux_put);
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
+
+#define WM8998_NG_SRC(name, base) \
+ SOC_SINGLE(name " NG HPOUTL Switch", base, 0, 1, 0), \
+ SOC_SINGLE(name " NG HPOUTR Switch", base, 1, 1, 0), \
+ SOC_SINGLE(name " NG LINEOUTL Switch", base, 2, 1, 0), \
+ SOC_SINGLE(name " NG LINEOUTR Switch", base, 3, 1, 0), \
+ SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \
+ SOC_SINGLE(name " NG SPKOUTL Switch", base, 6, 1, 0), \
+ SOC_SINGLE(name " NG SPKOUTR Switch", base, 7, 1, 0)
+
+static const struct snd_kcontrol_new wm8998_snd_controls[] = {
+SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]),
+SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]),
+
+SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
+
+SOC_SINGLE("IN1L HPF Switch", ARIZONA_IN1L_CONTROL,
+ ARIZONA_IN1L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN1R HPF Switch", ARIZONA_IN1R_CONTROL,
+ ARIZONA_IN1R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2 HPF Switch", ARIZONA_IN2L_CONTROL,
+ ARIZONA_IN2L_HPF_SHIFT, 1, 0),
+
+SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+ ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R,
+ ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+ ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+
+SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp),
+SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp),
+
+ARIZONA_GAINMUX_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE_SHIFT, 1, 0),
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE_SHIFT, 1, 0),
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE_SHIFT, 1, 0),
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE_SHIFT, 1, 0),
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+ 24, 0, eq_tlv),
+
+ARIZONA_GAINMUX_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+ ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1),
+SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1),
+SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1),
+SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]),
+SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]),
+SOC_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]),
+SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]),
+SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1),
+
+ARIZONA_MIXER_CONTROLS("HPOUTL", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUTR", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LINEOUTL", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LINEOUTR", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDATL", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDATR", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_DOUBLE_R("HPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("LINEOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("LINEOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE("SPKDAT Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+ ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+
+SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp),
+SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp),
+
+SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL,
+ ARIZONA_NGATE_ENA_SHIFT, 1, 0),
+SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL,
+ ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv),
+SOC_ENUM("Noise Gate Hold", arizona_ng_hold),
+
+WM8998_NG_SRC("HPOUTL", ARIZONA_NOISE_GATE_SELECT_1L),
+WM8998_NG_SRC("HPOUTR", ARIZONA_NOISE_GATE_SELECT_1R),
+WM8998_NG_SRC("LINEOUTL", ARIZONA_NOISE_GATE_SELECT_2L),
+WM8998_NG_SRC("LINEOUTR", ARIZONA_NOISE_GATE_SELECT_2R),
+WM8998_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L),
+WM8998_NG_SRC("SPKOUTL", ARIZONA_NOISE_GATE_SELECT_4L),
+WM8998_NG_SRC("SPKOUTR", ARIZONA_NOISE_GATE_SELECT_4R),
+WM8998_NG_SRC("SPKDATL", ARIZONA_NOISE_GATE_SELECT_5L),
+WM8998_NG_SRC("SPKDATR", ARIZONA_NOISE_GATE_SELECT_5R),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_GAINMUX_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
+
+ARIZONA_GAINMUX_CONTROLS("SPDIFTX1", ARIZONA_SPDIFTX1MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SPDIFTX2", ARIZONA_SPDIFTX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MUX_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDATL, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDATR, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(SPD1TX1, ARIZONA_SPDIFTX1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SPD1TX2, ARIZONA_SPDIFTX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT3, ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT4, ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC3, ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC4, ARIZONA_ISRC1DEC4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE);
+
+static const char * const wm8998_aec_loopback_texts[] = {
+ "HPOUTL", "HPOUTR", "LINEOUTL", "LINEOUTR", "EPOUT",
+ "SPKOUTL", "SPKOUTR", "SPKDATL", "SPKDATR",
+};
+
+static const unsigned int wm8998_aec_loopback_values[] = {
+ 0, 1, 2, 3, 4, 6, 7, 8, 9,
+};
+
+static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec1_loopback,
+ ARIZONA_DAC_AEC_CONTROL_1,
+ ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
+ wm8998_aec_loopback_texts,
+ wm8998_aec_loopback_values);
+
+static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec2_loopback,
+ ARIZONA_DAC_AEC_CONTROL_2,
+ ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
+ wm8998_aec_loopback_texts,
+ wm8998_aec_loopback_values);
+
+static const struct snd_kcontrol_new wm8998_aec_loopback_mux[] = {
+ SOC_DAPM_ENUM("AEC1 Loopback", wm8998_aec1_loopback),
+ SOC_DAPM_ENUM("AEC2 Loopback", wm8998_aec2_loopback),
+};
+
+static const struct snd_soc_dapm_widget wm8998_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1,
+ ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+ ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK,
+ ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK,
+ ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("HAPTICS"),
+
+SND_SOC_DAPM_INPUT("IN1AL"),
+SND_SOC_DAPM_INPUT("IN1AR"),
+SND_SOC_DAPM_INPUT("IN1BL"),
+SND_SOC_DAPM_INPUT("IN1BR"),
+SND_SOC_DAPM_INPUT("IN2A"),
+SND_SOC_DAPM_INPUT("IN2B"),
+
+SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &wm8998_in1mux[0]),
+SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &wm8998_in1mux[1]),
+SND_SOC_DAPM_MUX("IN2 Mux", SND_SOC_NOPM, 0, 0, &wm8998_in2mux),
+
+SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2 PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+ 0, NULL, 0, arizona_in_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+ ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+ ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+ 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_E("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+ NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+ NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+ NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+ NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+
+SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT3", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT4", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC3", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC4", ARIZONA_ISRC_1_CTRL_3,
+ ARIZONA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3,
+ ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_MUX("AEC1 Loopback", ARIZONA_DAC_AEC_CONTROL_1,
+ ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+ &wm8998_aec_loopback_mux[0]),
+
+SND_SOC_DAPM_MUX("AEC2 Loopback", ARIZONA_DAC_AEC_CONTROL_2,
+ ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+ &wm8998_aec_loopback_mux[1]),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+ ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+ ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0,
+ ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0,
+ ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0,
+ ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+ ARIZONA_SLIMRX4_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0,
+ ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+ ARIZONA_SLIMTX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+ ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+ ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+ ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+ ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
+ ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
+ ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_PGA("SPD1TX1", ARIZONA_SPD1_TX_CONTROL,
+ ARIZONA_SPD1_VAL1_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SPD1TX2", ARIZONA_SPD1_TX_CONTROL,
+ ARIZONA_SPD1_VAL2_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("SPD1", ARIZONA_SPD1_TX_CONTROL,
+ ARIZONA_SPD1_ENA_SHIFT, 0, NULL, 0),
+
+ARIZONA_MUX_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MUX_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MUX_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MUX_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MUX_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MUX_WIDGETS(DRC1R, "DRC1R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUTL"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUTR"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "LINEOUTL"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "LINEOUTR"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDATL, "SPKDATL"),
+ARIZONA_MIXER_WIDGETS(SPKDATR, "SPKDATR"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"),
+ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"),
+ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"),
+ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+ARIZONA_MUX_WIDGETS(SLIMTX1, "SLIMTX1"),
+ARIZONA_MUX_WIDGETS(SLIMTX2, "SLIMTX2"),
+ARIZONA_MUX_WIDGETS(SLIMTX3, "SLIMTX3"),
+ARIZONA_MUX_WIDGETS(SLIMTX4, "SLIMTX4"),
+ARIZONA_MUX_WIDGETS(SLIMTX5, "SLIMTX5"),
+ARIZONA_MUX_WIDGETS(SLIMTX6, "SLIMTX6"),
+
+ARIZONA_MUX_WIDGETS(SPD1TX1, "SPDIFTX1"),
+ARIZONA_MUX_WIDGETS(SPD1TX2, "SPDIFTX2"),
+
+ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"),
+
+ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"),
+
+ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"),
+ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"),
+ARIZONA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"),
+ARIZONA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"),
+
+ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
+
+ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
+ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
+
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("LINEOUTL"),
+SND_SOC_DAPM_OUTPUT("LINEOUTR"),
+SND_SOC_DAPM_OUTPUT("EPOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDATL"),
+SND_SOC_DAPM_OUTPUT("SPKDATR"),
+SND_SOC_DAPM_OUTPUT("SPDIF"),
+
+SND_SOC_DAPM_OUTPUT("MICSUPP"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name) \
+ { name, "Tone Generator 1", "Tone Generator 1" }, \
+ { name, "Tone Generator 2", "Tone Generator 2" }, \
+ { name, "Haptics", "HAPTICS" }, \
+ { name, "AEC", "AEC1 Loopback" }, \
+ { name, "AEC2", "AEC2 Loopback" }, \
+ { name, "IN1L", "IN1L PGA" }, \
+ { name, "IN1R", "IN1R PGA" }, \
+ { name, "IN2L", "IN2 PGA" }, \
+ { name, "AIF1RX1", "AIF1RX1" }, \
+ { name, "AIF1RX2", "AIF1RX2" }, \
+ { name, "AIF1RX3", "AIF1RX3" }, \
+ { name, "AIF1RX4", "AIF1RX4" }, \
+ { name, "AIF1RX5", "AIF1RX5" }, \
+ { name, "AIF1RX6", "AIF1RX6" }, \
+ { name, "AIF2RX1", "AIF2RX1" }, \
+ { name, "AIF2RX2", "AIF2RX2" }, \
+ { name, "AIF2RX3", "AIF2RX3" }, \
+ { name, "AIF2RX4", "AIF2RX4" }, \
+ { name, "AIF2RX5", "AIF2RX5" }, \
+ { name, "AIF2RX6", "AIF2RX6" }, \
+ { name, "AIF3RX1", "AIF3RX1" }, \
+ { name, "AIF3RX2", "AIF3RX2" }, \
+ { name, "SLIMRX1", "SLIMRX1" }, \
+ { name, "SLIMRX2", "SLIMRX2" }, \
+ { name, "SLIMRX3", "SLIMRX3" }, \
+ { name, "SLIMRX4", "SLIMRX4" }, \
+ { name, "EQ1", "EQ1" }, \
+ { name, "EQ2", "EQ2" }, \
+ { name, "EQ3", "EQ3" }, \
+ { name, "EQ4", "EQ4" }, \
+ { name, "DRC1L", "DRC1L" }, \
+ { name, "DRC1R", "DRC1R" }, \
+ { name, "LHPF1", "LHPF1" }, \
+ { name, "LHPF2", "LHPF2" }, \
+ { name, "LHPF3", "LHPF3" }, \
+ { name, "LHPF4", "LHPF4" }, \
+ { name, "ASRC1L", "ASRC1L" }, \
+ { name, "ASRC1R", "ASRC1R" }, \
+ { name, "ASRC2L", "ASRC2L" }, \
+ { name, "ASRC2R", "ASRC2R" }, \
+ { name, "ISRC1DEC1", "ISRC1DEC1" }, \
+ { name, "ISRC1DEC2", "ISRC1DEC2" }, \
+ { name, "ISRC1DEC3", "ISRC1DEC3" }, \
+ { name, "ISRC1DEC4", "ISRC1DEC4" }, \
+ { name, "ISRC1INT1", "ISRC1INT1" }, \
+ { name, "ISRC1INT2", "ISRC1INT2" }, \
+ { name, "ISRC1INT3", "ISRC1INT3" }, \
+ { name, "ISRC1INT4", "ISRC1INT4" }, \
+ { name, "ISRC2DEC1", "ISRC2DEC1" }, \
+ { name, "ISRC2DEC2", "ISRC2DEC2" }, \
+ { name, "ISRC2INT1", "ISRC2INT1" }, \
+ { name, "ISRC2INT2", "ISRC2INT2" }
+
+static const struct snd_soc_dapm_route wm8998_dapm_routes[] = {
+ { "AIF2 Capture", NULL, "DBVDD2" },
+ { "AIF2 Playback", NULL, "DBVDD2" },
+
+ { "AIF3 Capture", NULL, "DBVDD3" },
+ { "AIF3 Playback", NULL, "DBVDD3" },
+
+ { "OUT1L", NULL, "CPVDD" },
+ { "OUT1R", NULL, "CPVDD" },
+ { "OUT2L", NULL, "CPVDD" },
+ { "OUT2R", NULL, "CPVDD" },
+ { "OUT3", NULL, "CPVDD" },
+
+ { "OUT4L", NULL, "SPKVDDL" },
+ { "OUT4R", NULL, "SPKVDDR" },
+
+ { "OUT1L", NULL, "SYSCLK" },
+ { "OUT1R", NULL, "SYSCLK" },
+ { "OUT2L", NULL, "SYSCLK" },
+ { "OUT2R", NULL, "SYSCLK" },
+ { "OUT3", NULL, "SYSCLK" },
+ { "OUT4L", NULL, "SYSCLK" },
+ { "OUT4R", NULL, "SYSCLK" },
+ { "OUT5L", NULL, "SYSCLK" },
+ { "OUT5R", NULL, "SYSCLK" },
+
+ { "IN1AL", NULL, "SYSCLK" },
+ { "IN1AR", NULL, "SYSCLK" },
+ { "IN1BL", NULL, "SYSCLK" },
+ { "IN1BR", NULL, "SYSCLK" },
+ { "IN2A", NULL, "SYSCLK" },
+ { "IN2B", NULL, "SYSCLK" },
+
+ { "SPD1", NULL, "SYSCLK" },
+ { "SPD1", NULL, "SPD1TX1" },
+ { "SPD1", NULL, "SPD1TX2" },
+
+ { "MICBIAS1", NULL, "MICVDD" },
+ { "MICBIAS2", NULL, "MICVDD" },
+ { "MICBIAS3", NULL, "MICVDD" },
+
+ { "Tone Generator 1", NULL, "SYSCLK" },
+ { "Tone Generator 2", NULL, "SYSCLK" },
+
+ { "Tone Generator 1", NULL, "TONE" },
+ { "Tone Generator 2", NULL, "TONE" },
+
+ { "AIF1 Capture", NULL, "AIF1TX1" },
+ { "AIF1 Capture", NULL, "AIF1TX2" },
+ { "AIF1 Capture", NULL, "AIF1TX3" },
+ { "AIF1 Capture", NULL, "AIF1TX4" },
+ { "AIF1 Capture", NULL, "AIF1TX5" },
+ { "AIF1 Capture", NULL, "AIF1TX6" },
+
+ { "AIF1RX1", NULL, "AIF1 Playback" },
+ { "AIF1RX2", NULL, "AIF1 Playback" },
+ { "AIF1RX3", NULL, "AIF1 Playback" },
+ { "AIF1RX4", NULL, "AIF1 Playback" },
+ { "AIF1RX5", NULL, "AIF1 Playback" },
+ { "AIF1RX6", NULL, "AIF1 Playback" },
+
+ { "AIF2 Capture", NULL, "AIF2TX1" },
+ { "AIF2 Capture", NULL, "AIF2TX2" },
+ { "AIF2 Capture", NULL, "AIF2TX3" },
+ { "AIF2 Capture", NULL, "AIF2TX4" },
+ { "AIF2 Capture", NULL, "AIF2TX5" },
+ { "AIF2 Capture", NULL, "AIF2TX6" },
+
+ { "AIF2RX1", NULL, "AIF2 Playback" },
+ { "AIF2RX2", NULL, "AIF2 Playback" },
+ { "AIF2RX3", NULL, "AIF2 Playback" },
+ { "AIF2RX4", NULL, "AIF2 Playback" },
+ { "AIF2RX5", NULL, "AIF2 Playback" },
+ { "AIF2RX6", NULL, "AIF2 Playback" },
+
+ { "AIF3 Capture", NULL, "AIF3TX1" },
+ { "AIF3 Capture", NULL, "AIF3TX2" },
+
+ { "AIF3RX1", NULL, "AIF3 Playback" },
+ { "AIF3RX2", NULL, "AIF3 Playback" },
+
+ { "Slim1 Capture", NULL, "SLIMTX1" },
+ { "Slim1 Capture", NULL, "SLIMTX2" },
+ { "Slim1 Capture", NULL, "SLIMTX3" },
+ { "Slim1 Capture", NULL, "SLIMTX4" },
+
+ { "Slim2 Capture", NULL, "SLIMTX5" },
+ { "Slim2 Capture", NULL, "SLIMTX6" },
+
+ { "SLIMRX1", NULL, "Slim1 Playback" },
+ { "SLIMRX2", NULL, "Slim1 Playback" },
+
+ { "SLIMRX3", NULL, "Slim2 Playback" },
+ { "SLIMRX4", NULL, "Slim2 Playback" },
+
+ { "AIF1 Playback", NULL, "SYSCLK" },
+ { "AIF2 Playback", NULL, "SYSCLK" },
+ { "AIF3 Playback", NULL, "SYSCLK" },
+ { "Slim1 Playback", NULL, "SYSCLK" },
+ { "Slim2 Playback", NULL, "SYSCLK" },
+
+ { "AIF1 Capture", NULL, "SYSCLK" },
+ { "AIF2 Capture", NULL, "SYSCLK" },
+ { "AIF3 Capture", NULL, "SYSCLK" },
+ { "Slim1 Capture", NULL, "SYSCLK" },
+ { "Slim2 Capture", NULL, "SYSCLK" },
+
+ { "IN1L Mux", "A", "IN1AL" },
+ { "IN1R Mux", "A", "IN1AR" },
+ { "IN1L Mux", "B", "IN1BL" },
+ { "IN1R Mux", "B", "IN1BR" },
+
+ { "IN2 Mux", "A", "IN2A" },
+ { "IN2 Mux", "B", "IN2B" },
+
+ { "IN1L PGA", NULL, "IN1L Mux" },
+ { "IN1R PGA", NULL, "IN1R Mux" },
+ { "IN2 PGA", NULL, "IN2 Mux" },
+
+ ARIZONA_MIXER_ROUTES("OUT1L", "HPOUTL"),
+ ARIZONA_MIXER_ROUTES("OUT1R", "HPOUTR"),
+ ARIZONA_MIXER_ROUTES("OUT2L", "LINEOUTL"),
+ ARIZONA_MIXER_ROUTES("OUT2R", "LINEOUTR"),
+ ARIZONA_MIXER_ROUTES("OUT3", "EPOUT"),
+
+ ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+ ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+ ARIZONA_MIXER_ROUTES("OUT5L", "SPKDATL"),
+ ARIZONA_MIXER_ROUTES("OUT5R", "SPKDATR"),
+
+ ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+ ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+ ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+ ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+ ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+ ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+ ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+ ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+
+ ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+ ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+ ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"),
+ ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"),
+ ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"),
+ ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"),
+
+ ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+ ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+ ARIZONA_MUX_ROUTES("SLIMTX1", "SLIMTX1"),
+ ARIZONA_MUX_ROUTES("SLIMTX2", "SLIMTX2"),
+ ARIZONA_MUX_ROUTES("SLIMTX3", "SLIMTX3"),
+ ARIZONA_MUX_ROUTES("SLIMTX4", "SLIMTX4"),
+ ARIZONA_MUX_ROUTES("SLIMTX5", "SLIMTX5"),
+ ARIZONA_MUX_ROUTES("SLIMTX6", "SLIMTX6"),
+
+ ARIZONA_MUX_ROUTES("SPD1TX1", "SPDIFTX1"),
+ ARIZONA_MUX_ROUTES("SPD1TX2", "SPDIFTX2"),
+
+ ARIZONA_MUX_ROUTES("EQ1", "EQ1"),
+ ARIZONA_MUX_ROUTES("EQ2", "EQ2"),
+ ARIZONA_MUX_ROUTES("EQ3", "EQ3"),
+ ARIZONA_MUX_ROUTES("EQ4", "EQ4"),
+
+ ARIZONA_MUX_ROUTES("DRC1L", "DRC1L"),
+ ARIZONA_MUX_ROUTES("DRC1R", "DRC1R"),
+
+ ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+ ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+ ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+ ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+ ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"),
+ ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"),
+ ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
+ ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
+
+ ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+ ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"),
+ ARIZONA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"),
+ ARIZONA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"),
+
+ ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+ ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
+ ARIZONA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"),
+ ARIZONA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"),
+
+ ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+ ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+
+ ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+ ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
+
+ { "AEC1 Loopback", "HPOUTL", "OUT1L" },
+ { "AEC1 Loopback", "HPOUTR", "OUT1R" },
+ { "AEC2 Loopback", "HPOUTL", "OUT1L" },
+ { "AEC2 Loopback", "HPOUTR", "OUT1R" },
+ { "HPOUTL", NULL, "OUT1L" },
+ { "HPOUTR", NULL, "OUT1R" },
+
+ { "AEC1 Loopback", "LINEOUTL", "OUT2L" },
+ { "AEC1 Loopback", "LINEOUTR", "OUT2R" },
+ { "AEC2 Loopback", "LINEOUTL", "OUT2L" },
+ { "AEC2 Loopback", "LINEOUTR", "OUT2R" },
+ { "LINEOUTL", NULL, "OUT2L" },
+ { "LINEOUTR", NULL, "OUT2R" },
+
+ { "AEC1 Loopback", "EPOUT", "OUT3" },
+ { "AEC2 Loopback", "EPOUT", "OUT3" },
+ { "EPOUT", NULL, "OUT3" },
+
+ { "AEC1 Loopback", "SPKOUTL", "OUT4L" },
+ { "AEC2 Loopback", "SPKOUTL", "OUT4L" },
+ { "SPKOUTLN", NULL, "OUT4L" },
+ { "SPKOUTLP", NULL, "OUT4L" },
+
+ { "AEC1 Loopback", "SPKOUTR", "OUT4R" },
+ { "AEC2 Loopback", "SPKOUTR", "OUT4R" },
+ { "SPKOUTRN", NULL, "OUT4R" },
+ { "SPKOUTRP", NULL, "OUT4R" },
+
+ { "SPDIF", NULL, "SPD1" },
+
+ { "AEC1 Loopback", "SPKDATL", "OUT5L" },
+ { "AEC1 Loopback", "SPKDATR", "OUT5R" },
+ { "AEC2 Loopback", "SPKDATL", "OUT5L" },
+ { "AEC2 Loopback", "SPKDATR", "OUT5R" },
+ { "SPKDATL", NULL, "OUT5L" },
+ { "SPKDATR", NULL, "OUT5R" },
+
+ { "MICSUPP", NULL, "SYSCLK" },
+
+ { "DRC1 Signal Activity", NULL, "DRC1L" },
+ { "DRC1 Signal Activity", NULL, "DRC1R" },
+};
+
+#define WM8998_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM8998_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver wm8998_dai[] = {
+ {
+ .name = "wm8998-aif1",
+ .id = 1,
+ .base = ARIZONA_AIF1_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ .symmetric_samplebits = 1,
+ },
+ {
+ .name = "wm8998-aif2",
+ .id = 2,
+ .base = ARIZONA_AIF2_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ .symmetric_samplebits = 1,
+ },
+ {
+ .name = "wm8998-aif3",
+ .id = 3,
+ .base = ARIZONA_AIF3_BCLK_CTRL,
+ .playback = {
+ .stream_name = "AIF3 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF3 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .ops = &arizona_dai_ops,
+ .symmetric_rates = 1,
+ .symmetric_samplebits = 1,
+ },
+ {
+ .name = "wm8998-slim1",
+ .id = 4,
+ .playback = {
+ .stream_name = "Slim1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Slim1 Capture",
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .ops = &arizona_simple_dai_ops,
+ },
+ {
+ .name = "wm8998-slim2",
+ .id = 5,
+ .playback = {
+ .stream_name = "Slim2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Slim2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8998_RATES,
+ .formats = WM8998_FORMATS,
+ },
+ .ops = &arizona_simple_dai_ops,
+ },
+};
+
+static int wm8998_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec);
+
+ switch (fll_id) {
+ case WM8998_FLL1:
+ return arizona_set_fll(&wm8998->fll[0], source, Fref, Fout);
+ case WM8998_FLL2:
+ return arizona_set_fll(&wm8998->fll[1], source, Fref, Fout);
+ case WM8998_FLL1_REFCLK:
+ return arizona_set_fll_refclk(&wm8998->fll[0], source, Fref,
+ Fout);
+ case WM8998_FLL2_REFCLK:
+ return arizona_set_fll_refclk(&wm8998->fll[1], source, Fref,
+ Fout);
+ default:
+ return -EINVAL;
+ }
+}
+
+static int wm8998_codec_probe(struct snd_soc_codec *codec)
+{
+ struct wm8998_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+
+ priv->core.arizona->dapm = dapm;
+
+ arizona_init_spk(codec);
+ arizona_init_gpio(codec);
+
+ snd_soc_dapm_disable_pin(dapm, "HAPTICS");
+
+ return 0;
+}
+
+static int wm8998_codec_remove(struct snd_soc_codec *codec)
+{
+ struct wm8998_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ priv->core.arizona->dapm = NULL;
+
+ return 0;
+}
+
+#define WM8998_DIG_VU 0x0200
+
+static unsigned int wm8998_digital_vu[] = {
+ ARIZONA_DAC_DIGITAL_VOLUME_1L,
+ ARIZONA_DAC_DIGITAL_VOLUME_1R,
+ ARIZONA_DAC_DIGITAL_VOLUME_2L,
+ ARIZONA_DAC_DIGITAL_VOLUME_2R,
+ ARIZONA_DAC_DIGITAL_VOLUME_3L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4L,
+ ARIZONA_DAC_DIGITAL_VOLUME_4R,
+ ARIZONA_DAC_DIGITAL_VOLUME_5L,
+ ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct regmap *wm8998_get_regmap(struct device *dev)
+{
+ struct wm8998_priv *priv = dev_get_drvdata(dev);
+
+ return priv->core.arizona->regmap;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8998 = {
+ .probe = wm8998_codec_probe,
+ .remove = wm8998_codec_remove,
+ .get_regmap = wm8998_get_regmap,
+
+ .idle_bias_off = true,
+
+ .set_sysclk = arizona_set_sysclk,
+ .set_pll = wm8998_set_fll,
+
+ .controls = wm8998_snd_controls,
+ .num_controls = ARRAY_SIZE(wm8998_snd_controls),
+ .dapm_widgets = wm8998_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8998_dapm_widgets),
+ .dapm_routes = wm8998_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm8998_dapm_routes),
+};
+
+static int wm8998_probe(struct platform_device *pdev)
+{
+ struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+ struct wm8998_priv *wm8998;
+ int i;
+
+ wm8998 = devm_kzalloc(&pdev->dev, sizeof(struct wm8998_priv),
+ GFP_KERNEL);
+ if (!wm8998)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, wm8998);
+
+ wm8998->core.arizona = arizona;
+ wm8998->core.num_inputs = 3; /* IN1L, IN1R, IN2 */
+
+ for (i = 0; i < ARRAY_SIZE(wm8998->fll); i++)
+ wm8998->fll[i].vco_mult = 1;
+
+ arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+ &wm8998->fll[0]);
+ arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+ ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+ &wm8998->fll[1]);
+
+ for (i = 0; i < ARRAY_SIZE(wm8998_dai); i++)
+ arizona_init_dai(&wm8998->core, i);
+
+ /* Latch volume update bits */
+ for (i = 0; i < ARRAY_SIZE(wm8998_digital_vu); i++)
+ regmap_update_bits(arizona->regmap, wm8998_digital_vu[i],
+ WM8998_DIG_VU, WM8998_DIG_VU);
+
+ pm_runtime_enable(&pdev->dev);
+ pm_runtime_idle(&pdev->dev);
+
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8998,
+ wm8998_dai, ARRAY_SIZE(wm8998_dai));
+}
+
+static int wm8998_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver wm8998_codec_driver = {
+ .driver = {
+ .name = "wm8998-codec",
+ },
+ .probe = wm8998_probe,
+ .remove = wm8998_remove,
+};
+
+module_platform_driver(wm8998_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM8998 driver");
+MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:wm8998-codec");
diff --git a/sound/soc/codecs/wm8998.h b/sound/soc/codecs/wm8998.h
new file mode 100644
index 0000000..1e86472
--- /dev/null
+++ b/sound/soc/codecs/wm8998.h
@@ -0,0 +1,23 @@
+/*
+ * wm8998.h -- ALSA SoC Audio driver for WM8998 codecs
+ *
+ * Copyright 2015 Cirrus Logic, Inc.
+ *
+ * Author: Richard Fitzgerald <rf@opensource.wolfsonmicro.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 _WM8998_H
+#define _WM8998_H
+
+#include "arizona.h"
+
+#define WM8998_FLL1 1
+#define WM8998_FLL2 2
+#define WM8998_FLL1_REFCLK 3
+#define WM8998_FLL2_REFCLK 4
+
+#endif
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7d45d98..4495a40 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -80,12 +80,13 @@
/* McASP specific data */
int tdm_slots;
+ u32 tdm_mask[2];
+ int slot_width;
u8 op_mode;
u8 num_serializer;
u8 *serial_dir;
u8 version;
u8 bclk_div;
- u16 bclk_lrclk_ratio;
int streams;
u32 irq_request[2];
int dma_request[2];
@@ -556,8 +557,21 @@
mcasp->bclk_div = div;
break;
- case 2: /* BCLK/LRCLK ratio */
- mcasp->bclk_lrclk_ratio = div;
+ case 2: /*
+ * BCLK/LRCLK ratio descries how many bit-clock cycles
+ * fit into one frame. The clock ratio is given for a
+ * full period of data (for I2S format both left and
+ * right channels), so it has to be divided by number
+ * of tdm-slots (for I2S - divided by 2).
+ * Instead of storing this ratio, we calculate a new
+ * tdm_slot width by dividing the the ratio by the
+ * number of configured tdm slots.
+ */
+ mcasp->slot_width = div / mcasp->tdm_slots;
+ if (div % mcasp->tdm_slots)
+ dev_warn(mcasp->dev,
+ "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots",
+ __func__, div, mcasp->tdm_slots);
break;
default:
@@ -596,12 +610,92 @@
return 0;
}
+/* All serializers must have equal number of channels */
+static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
+ int serializers)
+{
+ struct snd_pcm_hw_constraint_list *cl = &mcasp->chconstr[stream];
+ unsigned int *list = (unsigned int *) cl->list;
+ int slots = mcasp->tdm_slots;
+ int i, count = 0;
+
+ if (mcasp->tdm_mask[stream])
+ slots = hweight32(mcasp->tdm_mask[stream]);
+
+ for (i = 2; i <= slots; i++)
+ list[count++] = i;
+
+ for (i = 2; i <= serializers; i++)
+ list[count++] = i*slots;
+
+ cl->count = count;
+
+ return 0;
+}
+
+static int davinci_mcasp_set_ch_constraints(struct davinci_mcasp *mcasp)
+{
+ int rx_serializers = 0, tx_serializers = 0, ret, i;
+
+ for (i = 0; i < mcasp->num_serializer; i++)
+ if (mcasp->serial_dir[i] == TX_MODE)
+ tx_serializers++;
+ else if (mcasp->serial_dir[i] == RX_MODE)
+ rx_serializers++;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_PLAYBACK,
+ tx_serializers);
+ if (ret)
+ return ret;
+
+ ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_CAPTURE,
+ rx_serializers);
+
+ return ret;
+}
+
+
+static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask,
+ unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(mcasp->dev,
+ "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n",
+ __func__, tx_mask, rx_mask, slots, slot_width);
+
+ if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
+ dev_err(mcasp->dev,
+ "Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n",
+ tx_mask, rx_mask, slots);
+ return -EINVAL;
+ }
+
+ if (slot_width &&
+ (slot_width < 8 || slot_width > 32 || slot_width % 4 != 0)) {
+ dev_err(mcasp->dev, "%s: Unsupported slot_width %d\n",
+ __func__, slot_width);
+ return -EINVAL;
+ }
+
+ mcasp->tdm_slots = slots;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
+ mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+ mcasp->slot_width = slot_width;
+
+ return davinci_mcasp_set_ch_constraints(mcasp);
+}
+
static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
- int word_length)
+ int sample_width)
{
u32 fmt;
- u32 tx_rotate = (word_length / 4) & 0x7;
- u32 mask = (1ULL << word_length) - 1;
+ u32 tx_rotate = (sample_width / 4) & 0x7;
+ u32 mask = (1ULL << sample_width) - 1;
+ u32 slot_width = sample_width;
+
/*
* For captured data we should not rotate, inversion and masking is
* enoguh to get the data to the right position:
@@ -614,28 +708,23 @@
u32 rx_rotate = 0;
/*
- * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
- * callback, take it into account here. That allows us to for example
- * send 32 bits per channel to the codec, while only 16 of them carry
- * audio payload.
- * The clock ratio is given for a full period of data (for I2S format
- * both left and right channels), so it has to be divided by number of
- * tdm-slots (for I2S - divided by 2).
+ * Setting the tdm slot width either with set_clkdiv() or
+ * set_tdm_slot() allows us to for example send 32 bits per
+ * channel to the codec, while only 16 of them carry audio
+ * payload.
*/
- if (mcasp->bclk_lrclk_ratio) {
- u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
-
+ if (mcasp->slot_width) {
/*
- * When we have more bclk then it is needed for the data, we
- * need to use the rotation to move the received samples to have
- * correct alignment.
+ * When we have more bclk then it is needed for the
+ * data, we need to use the rotation to move the
+ * received samples to have correct alignment.
*/
- rx_rotate = (slot_length - word_length) / 4;
- word_length = slot_length;
+ slot_width = mcasp->slot_width;
+ rx_rotate = (slot_width - sample_width) / 4;
}
/* mapping of the XSSZ bit-field as described in the datasheet */
- fmt = (word_length >> 1) - 1;
+ fmt = (slot_width >> 1) - 1;
if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt),
@@ -776,33 +865,50 @@
/*
* If more than one serializer is needed, then use them with
- * their specified tdm_slots count. Otherwise, one serializer
- * can cope with the transaction using as many slots as channels
- * in the stream, requires channels symmetry
+ * all the specified tdm_slots. Otherwise, one serializer can
+ * cope with the transaction using just as many slots as there
+ * are channels in the stream.
*/
- active_serializers = (channels + total_slots - 1) / total_slots;
- if (active_serializers == 1)
- active_slots = channels;
- else
- active_slots = total_slots;
+ if (mcasp->tdm_mask[stream]) {
+ active_slots = hweight32(mcasp->tdm_mask[stream]);
+ active_serializers = (channels + active_slots - 1) /
+ active_slots;
+ if (active_serializers == 1) {
+ active_slots = channels;
+ for (i = 0; i < total_slots; i++) {
+ if ((1 << i) & mcasp->tdm_mask[stream]) {
+ mask |= (1 << i);
+ if (--active_slots <= 0)
+ break;
+ }
+ }
+ }
+ } else {
+ active_serializers = (channels + total_slots - 1) / total_slots;
+ if (active_serializers == 1)
+ active_slots = channels;
+ else
+ active_slots = total_slots;
- for (i = 0; i < active_slots; i++)
- mask |= (1 << i);
-
+ for (i = 0; i < active_slots; i++)
+ mask |= (1 << i);
+ }
mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
if (!mcasp->dat_port)
busel = TXSEL;
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
- FSXMOD(total_slots), FSXMOD(0x1FF));
-
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
- FSRMOD(total_slots), FSRMOD(0x1FF));
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(total_slots), FSXMOD(0x1FF));
+ } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+ FSRMOD(total_slots), FSRMOD(0x1FF));
+ }
return 0;
}
@@ -922,6 +1028,9 @@
int sbits = params_width(params);
int ppm, div;
+ if (mcasp->slot_width)
+ sbits = mcasp->slot_width;
+
div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots,
&ppm);
if (ppm)
@@ -1027,6 +1136,9 @@
struct snd_interval range;
int i;
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
snd_interval_any(&range);
range.empty = 1;
@@ -1069,10 +1181,14 @@
for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
if (snd_mask_test(fmt, i)) {
- uint bclk_freq = snd_pcm_format_width(i)*slots*rate;
+ uint sbits = snd_pcm_format_width(i);
int ppm;
- davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+ if (rd->mcasp->slot_width)
+ sbits = rd->mcasp->slot_width;
+
+ davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
+ &ppm);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
snd_mask_set(&nfmt, i);
count++;
@@ -1094,6 +1210,10 @@
&mcasp->ruledata[substream->stream];
u32 max_channels = 0;
int i, dir;
+ int tdm_slots = mcasp->tdm_slots;
+
+ if (mcasp->tdm_mask[substream->stream])
+ tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
mcasp->substreams[substream->stream] = substream;
@@ -1114,7 +1234,7 @@
max_channels++;
}
ruledata->serializers = max_channels;
- max_channels *= mcasp->tdm_slots;
+ max_channels *= tdm_slots;
/*
* If the already active stream has less channels than the calculated
* limnit based on the seirializers * tdm_slots, we need to use that as
@@ -1124,15 +1244,25 @@
*/
if (mcasp->channels && mcasp->channels < max_channels)
max_channels = mcasp->channels;
+ /*
+ * But we can always allow channels upto the amount of
+ * the available tdm_slots.
+ */
+ if (max_channels < tdm_slots)
+ max_channels = tdm_slots;
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
2, max_channels);
- if (mcasp->chconstr[substream->stream].count)
- snd_pcm_hw_constraint_list(substream->runtime,
- 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- &mcasp->chconstr[substream->stream]);
+ snd_pcm_hw_constraint_list(substream->runtime,
+ 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &mcasp->chconstr[substream->stream]);
+
+ if (mcasp->slot_width)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ 8, mcasp->slot_width);
/*
* If we rely on implicit BCLK divider setting we should
@@ -1184,6 +1314,7 @@
.set_fmt = davinci_mcasp_set_dai_fmt,
.set_clkdiv = davinci_mcasp_set_clkdiv,
.set_sysclk = davinci_mcasp_set_sysclk,
+ .set_tdm_slot = davinci_mcasp_set_tdm_slot,
};
static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
@@ -1514,59 +1645,6 @@
return pdata;
}
-/* All serializers must have equal number of channels */
-static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp,
- struct snd_pcm_hw_constraint_list *cl,
- int serializers)
-{
- unsigned int *list;
- int i, count = 0;
-
- if (serializers <= 1)
- return 0;
-
- list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
- (mcasp->tdm_slots + serializers - 2),
- GFP_KERNEL);
- if (!list)
- return -ENOMEM;
-
- for (i = 2; i <= mcasp->tdm_slots; i++)
- list[count++] = i;
-
- for (i = 2; i <= serializers; i++)
- list[count++] = i*mcasp->tdm_slots;
-
- cl->count = count;
- cl->list = list;
-
- return 0;
-}
-
-
-static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp)
-{
- int rx_serializers = 0, tx_serializers = 0, ret, i;
-
- for (i = 0; i < mcasp->num_serializer; i++)
- if (mcasp->serial_dir[i] == TX_MODE)
- tx_serializers++;
- else if (mcasp->serial_dir[i] == RX_MODE)
- rx_serializers++;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_PLAYBACK],
- tx_serializers);
- if (ret)
- return ret;
-
- ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
- SNDRV_PCM_STREAM_CAPTURE],
- rx_serializers);
-
- return ret;
-}
-
enum {
PCM_EDMA,
PCM_SDMA,
@@ -1783,7 +1861,28 @@
mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
}
- ret = davinci_mcasp_init_ch_constraints(mcasp);
+ /* Allocate memory for long enough list for all possible
+ * scenarios. Maximum number tdm slots is 32 and there cannot
+ * be more serializers than given in the configuration. The
+ * serializer directions could be taken into account, but it
+ * would make code much more complex and save only couple of
+ * bytes.
+ */
+ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
+ devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+ (32 + mcasp->num_serializer - 2),
+ GFP_KERNEL);
+
+ if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
+ !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list)
+ return -ENOMEM;
+
+ ret = davinci_mcasp_set_ch_constraints(mcasp);
if (ret)
goto err;
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index ba34252..6e6a70c 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -282,23 +282,25 @@
config->sample_rate = params_rate(params);
- if (dev->i2s_clk_cfg) {
- ret = dev->i2s_clk_cfg(config);
- if (ret < 0) {
- dev_err(dev->dev, "runtime audio clk config fail\n");
- return ret;
- }
- } else {
- u32 bitclk = config->sample_rate * config->data_width * 2;
+ if (dev->capability & DW_I2S_MASTER) {
+ if (dev->i2s_clk_cfg) {
+ ret = dev->i2s_clk_cfg(config);
+ if (ret < 0) {
+ dev_err(dev->dev, "runtime audio clk config fail\n");
+ return ret;
+ }
+ } else {
+ u32 bitclk = config->sample_rate *
+ config->data_width * 2;
- ret = clk_set_rate(dev->clk, bitclk);
- if (ret) {
- dev_err(dev->dev, "Can't set I2S clock rate: %d\n",
- ret);
- return ret;
+ ret = clk_set_rate(dev->clk, bitclk);
+ if (ret) {
+ dev_err(dev->dev, "Can't set I2S clock rate: %d\n",
+ ret);
+ return ret;
+ }
}
}
-
return 0;
}
@@ -348,12 +350,43 @@
return ret;
}
+static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ if (dev->capability & DW_I2S_SLAVE)
+ ret = 0;
+ else
+ ret = -EINVAL;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ if (dev->capability & DW_I2S_MASTER)
+ ret = 0;
+ else
+ ret = -EINVAL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ ret = -EINVAL;
+ break;
+ default:
+ dev_dbg(dev->dev, "dwc : Invalid master/slave format\n");
+ ret = -EINVAL;
+ break;
+ }
+ return ret;
+}
+
static struct snd_soc_dai_ops dw_i2s_dai_ops = {
.startup = dw_i2s_startup,
.shutdown = dw_i2s_shutdown,
.hw_params = dw_i2s_hw_params,
.prepare = dw_i2s_prepare,
.trigger = dw_i2s_trigger,
+ .set_fmt = dw_i2s_set_fmt,
};
static const struct snd_soc_component_driver dw_i2s_component = {
@@ -366,7 +399,8 @@
{
struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
- clk_disable(dev->clk);
+ if (dev->capability & DW_I2S_MASTER)
+ clk_disable(dev->clk);
return 0;
}
@@ -374,7 +408,8 @@
{
struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
- clk_enable(dev->clk);
+ if (dev->capability & DW_I2S_MASTER)
+ clk_enable(dev->clk);
return 0;
}
@@ -452,6 +487,14 @@
dw_i2s_dai->capture.rates = rates;
}
+ if (COMP1_MODE_EN(comp1)) {
+ dev_dbg(dev->dev, "designware: i2s master mode supported\n");
+ dev->capability |= DW_I2S_MASTER;
+ } else {
+ dev_dbg(dev->dev, "designware: i2s slave mode supported\n");
+ dev->capability |= DW_I2S_SLAVE;
+ }
+
return 0;
}
@@ -538,6 +581,7 @@
struct resource *res;
int ret;
struct snd_soc_dai_driver *dw_i2s_dai;
+ const char *clk_id;
dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
if (!dev) {
@@ -559,33 +603,36 @@
return PTR_ERR(dev->i2s_base);
dev->dev = &pdev->dev;
+
if (pdata) {
- ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
- if (ret < 0)
- return ret;
-
dev->capability = pdata->cap;
- dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
- if (!dev->i2s_clk_cfg) {
- dev_err(&pdev->dev, "no clock configure method\n");
- return -ENODEV;
- }
-
- dev->clk = devm_clk_get(&pdev->dev, NULL);
+ clk_id = NULL;
+ ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
} else {
+ clk_id = "i2sclk";
ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res);
- if (ret < 0)
- return ret;
-
- dev->clk = devm_clk_get(&pdev->dev, "i2sclk");
}
- if (IS_ERR(dev->clk))
- return PTR_ERR(dev->clk);
-
- ret = clk_prepare_enable(dev->clk);
if (ret < 0)
return ret;
+ if (dev->capability & DW_I2S_MASTER) {
+ if (pdata) {
+ dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
+ if (!dev->i2s_clk_cfg) {
+ dev_err(&pdev->dev, "no clock configure method\n");
+ return -ENODEV;
+ }
+ }
+ dev->clk = devm_clk_get(&pdev->dev, clk_id);
+
+ if (IS_ERR(dev->clk))
+ return PTR_ERR(dev->clk);
+
+ ret = clk_prepare_enable(dev->clk);
+ if (ret < 0)
+ return ret;
+ }
+
dev_set_drvdata(&pdev->dev, dev);
ret = devm_snd_soc_register_component(&pdev->dev, &dw_i2s_component,
dw_i2s_dai, 1);
@@ -606,7 +653,8 @@
return 0;
err_clk_disable:
- clk_disable_unprepare(dev->clk);
+ if (dev->capability & DW_I2S_MASTER)
+ clk_disable_unprepare(dev->clk);
return ret;
}
@@ -614,7 +662,8 @@
{
struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
- clk_disable_unprepare(dev->clk);
+ if (dev->capability & DW_I2S_MASTER)
+ clk_disable_unprepare(dev->clk);
return 0;
}
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 96f55ae..1b05d1c 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -14,6 +14,9 @@
#include <linux/i2c.h>
#include <linux/module.h>
#include <linux/of_platform.h>
+#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
+#include <sound/ac97_codec.h>
+#endif
#include <sound/pcm_params.h>
#include <sound/soc.h>
@@ -115,6 +118,11 @@
SND_SOC_DAPM_MIC("DMIC", NULL),
};
+static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
+{
+ return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
+}
+
static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -133,7 +141,9 @@
* set_bias_level(), bypass the remaining settings in hw_params().
* Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
*/
- if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
+ if ((priv->card.set_bias_level &&
+ priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
+ fsl_asoc_card_is_ac97(priv))
return 0;
/* Specific configurations of DAIs starts from here */
@@ -300,7 +310,7 @@
ext_port--;
/*
- * Use asynchronous mode (6 wires) for all cases.
+ * Use asynchronous mode (6 wires) for all cases except AC97.
* If only 4 wires are needed, just set SSI into
* synchronous mode and enable 4 PADs in IOMUX.
*/
@@ -346,15 +356,30 @@
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
default:
- return -EINVAL;
+ if (!fsl_asoc_card_is_ac97(priv))
+ return -EINVAL;
+ }
+
+ if (fsl_asoc_card_is_ac97(priv)) {
+ int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR;
+ ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR;
}
/* Asynchronous mode can not be set along with RCLKDIR */
- ret = imx_audmux_v2_configure_port(int_port, 0,
- IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
- if (ret) {
- dev_err(dev, "audmux internal port setup failed\n");
- return ret;
+ if (!fsl_asoc_card_is_ac97(priv)) {
+ unsigned int pdcr =
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
+
+ ret = imx_audmux_v2_configure_port(int_port, 0,
+ pdcr);
+ if (ret) {
+ dev_err(dev, "audmux internal port setup failed\n");
+ return ret;
+ }
}
ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
@@ -364,11 +389,16 @@
return ret;
}
- ret = imx_audmux_v2_configure_port(ext_port, 0,
- IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
- if (ret) {
- dev_err(dev, "audmux external port setup failed\n");
- return ret;
+ if (!fsl_asoc_card_is_ac97(priv)) {
+ unsigned int pdcr =
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
+
+ ret = imx_audmux_v2_configure_port(ext_port, 0,
+ pdcr);
+ if (ret) {
+ dev_err(dev, "audmux external port setup failed\n");
+ return ret;
+ }
}
ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
@@ -389,6 +419,23 @@
struct device *dev = card->dev;
int ret;
+ if (fsl_asoc_card_is_ac97(priv)) {
+#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
+ struct snd_soc_codec *codec = card->rtd[0].codec;
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+ /*
+ * Use slots 3/4 for S/PDIF so SSI won't try to enable
+ * other slots and send some samples there
+ * due to SLOTREQ bits for S/PDIF received from codec
+ */
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
+ AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
+#endif
+
+ return 0;
+ }
+
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
if (ret) {
@@ -407,7 +454,6 @@
struct platform_device *cpu_pdev;
struct fsl_asoc_card_priv *priv;
struct i2c_client *codec_dev;
- struct clk *codec_clk;
const char *codec_dai_name;
u32 width;
int ret;
@@ -420,9 +466,8 @@
/* Give a chance to old DT binding */
if (!cpu_np)
cpu_np = of_parse_phandle(np, "ssi-controller", 0);
- codec_np = of_parse_phandle(np, "audio-codec", 0);
- if (!cpu_np || !codec_np) {
- dev_err(&pdev->dev, "phandle missing or invalid\n");
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
ret = -EINVAL;
goto fail;
}
@@ -434,22 +479,24 @@
goto fail;
}
- codec_dev = of_find_i2c_device_by_node(codec_np);
- if (!codec_dev) {
- dev_err(&pdev->dev, "failed to find codec platform device\n");
- ret = -EINVAL;
- goto fail;
- }
+ codec_np = of_parse_phandle(np, "audio-codec", 0);
+ if (codec_np)
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ else
+ codec_dev = NULL;
asrc_np = of_parse_phandle(np, "audio-asrc", 0);
if (asrc_np)
asrc_pdev = of_find_device_by_node(asrc_np);
/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
- codec_clk = clk_get(&codec_dev->dev, NULL);
- if (!IS_ERR(codec_clk)) {
- priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
- clk_put(codec_clk);
+ if (codec_dev) {
+ struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
+
+ if (!IS_ERR(codec_clk)) {
+ priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+ clk_put(codec_clk);
+ }
}
/* Default sample rate and format, will be updated in hw_params() */
@@ -486,12 +533,22 @@
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
+ codec_dai_name = "ac97-hifi";
+ priv->card.set_bias_level = NULL;
+ priv->dai_fmt = SND_SOC_DAIFMT_AC97;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
goto asrc_fail;
}
+ if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
+ dev_err(&pdev->dev, "failed to find codec device\n");
+ ret = -EINVAL;
+ goto asrc_fail;
+ }
+
/* Common settings for corresponding Freescale CPU DAI driver */
if (strstr(cpu_np->name, "ssi")) {
/* Only SSI needs to configure AUDMUX */
@@ -508,7 +565,9 @@
priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
}
- sprintf(priv->name, "%s-audio", codec_dev->name);
+ snprintf(priv->name, sizeof(priv->name), "%s-audio",
+ fsl_asoc_card_is_ac97(priv) ? "ac97" :
+ codec_dev->name);
/* Initialize sound card */
priv->pdev = pdev;
@@ -532,8 +591,26 @@
/* Normal DAI Link */
priv->dai_link[0].cpu_of_node = cpu_np;
- priv->dai_link[0].codec_of_node = codec_np;
priv->dai_link[0].codec_dai_name = codec_dai_name;
+
+ if (!fsl_asoc_card_is_ac97(priv))
+ priv->dai_link[0].codec_of_node = codec_np;
+ else {
+ u32 idx;
+
+ ret = of_property_read_u32(cpu_np, "cell-index", &idx);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "cannot get CPU index property\n");
+ goto asrc_fail;
+ }
+
+ priv->dai_link[0].codec_name =
+ devm_kasprintf(&pdev->dev, GFP_KERNEL,
+ "ac97-codec.%u",
+ (unsigned int)idx);
+ }
+
priv->dai_link[0].platform_of_node = cpu_np;
priv->dai_link[0].dai_fmt = priv->dai_fmt;
priv->card.num_links = 1;
@@ -544,6 +621,8 @@
priv->dai_link[1].platform_of_node = asrc_np;
priv->dai_link[2].codec_dai_name = codec_dai_name;
priv->dai_link[2].codec_of_node = codec_np;
+ priv->dai_link[2].codec_name =
+ priv->dai_link[0].codec_name;
priv->dai_link[2].cpu_of_node = cpu_np;
priv->dai_link[2].dai_fmt = priv->dai_fmt;
priv->card.num_links = 3;
@@ -579,20 +658,22 @@
asrc_fail:
of_node_put(asrc_np);
-fail:
of_node_put(codec_np);
+fail:
of_node_put(cpu_np);
return ret;
}
static const struct of_device_id fsl_asoc_card_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-ac97", },
{ .compatible = "fsl,imx-audio-cs42888", },
{ .compatible = "fsl,imx-audio-sgtl5000", },
{ .compatible = "fsl,imx-audio-wm8962", },
{ .compatible = "fsl,imx-audio-wm8960", },
{}
};
+MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
static struct platform_driver fsl_asoc_card_driver = {
.probe = fsl_asoc_card_probe,
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 837979e..59f234e 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -652,6 +652,24 @@
.name = "fsl-esai",
};
+static const struct reg_default fsl_esai_reg_defaults[] = {
+ {0x8, 0x00000000},
+ {0x10, 0x00000000},
+ {0x18, 0x00000000},
+ {0x98, 0x00000000},
+ {0xd0, 0x00000000},
+ {0xd4, 0x00000000},
+ {0xd8, 0x00000000},
+ {0xdc, 0x00000000},
+ {0xe0, 0x00000000},
+ {0xe4, 0x0000ffff},
+ {0xe8, 0x0000ffff},
+ {0xec, 0x0000ffff},
+ {0xf0, 0x0000ffff},
+ {0xf8, 0x00000000},
+ {0xfc, 0x00000000},
+};
+
static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -684,6 +702,31 @@
}
}
+static bool fsl_esai_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ESAI_ETDR:
+ case REG_ESAI_ERDR:
+ case REG_ESAI_ESR:
+ case REG_ESAI_TFSR:
+ case REG_ESAI_RFSR:
+ case REG_ESAI_TX0:
+ case REG_ESAI_TX1:
+ case REG_ESAI_TX2:
+ case REG_ESAI_TX3:
+ case REG_ESAI_TX4:
+ case REG_ESAI_TX5:
+ case REG_ESAI_RX0:
+ case REG_ESAI_RX1:
+ case REG_ESAI_RX2:
+ case REG_ESAI_RX3:
+ case REG_ESAI_SAISR:
+ return true;
+ default:
+ return false;
+ }
+}
+
static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -721,8 +764,12 @@
.val_bits = 32,
.max_register = REG_ESAI_PCRC,
+ .reg_defaults = fsl_esai_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(fsl_esai_reg_defaults),
.readable_reg = fsl_esai_readable_reg,
+ .volatile_reg = fsl_esai_volatile_reg,
.writeable_reg = fsl_esai_writeable_reg,
+ .cache_type = REGCACHE_RBTREE,
};
static int fsl_esai_probe(struct platform_device *pdev)
@@ -853,10 +900,51 @@
};
MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids);
+#ifdef CONFIG_PM_SLEEP
+static int fsl_esai_suspend(struct device *dev)
+{
+ struct fsl_esai *esai = dev_get_drvdata(dev);
+
+ regcache_cache_only(esai->regmap, true);
+ regcache_mark_dirty(esai->regmap);
+
+ return 0;
+}
+
+static int fsl_esai_resume(struct device *dev)
+{
+ struct fsl_esai *esai = dev_get_drvdata(dev);
+ int ret;
+
+ regcache_cache_only(esai->regmap, false);
+
+ /* FIFO reset for safety */
+ regmap_update_bits(esai->regmap, REG_ESAI_TFCR,
+ ESAI_xFCR_xFR, ESAI_xFCR_xFR);
+ regmap_update_bits(esai->regmap, REG_ESAI_RFCR,
+ ESAI_xFCR_xFR, ESAI_xFCR_xFR);
+
+ ret = regcache_sync(esai->regmap);
+ if (ret)
+ return ret;
+
+ /* FIFO reset done */
+ regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0);
+ regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0);
+
+ return 0;
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_esai_pm_ops = {
+ SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume)
+};
+
static struct platform_driver fsl_esai_driver = {
.probe = fsl_esai_probe,
.driver = {
.name = "fsl-esai-dai",
+ .pm = &fsl_esai_pm_ops,
.of_match_table = fsl_esai_dt_ids,
},
};
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a18fd92..a4435f5 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -27,13 +27,13 @@
#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\
FSL_SAI_CSR_FEIE)
-static u32 fsl_sai_rates[] = {
+static const unsigned int fsl_sai_rates[] = {
8000, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000, 64000,
88200, 96000, 176400, 192000
};
-static struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = {
+static const struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = {
.count = ARRAY_SIZE(fsl_sai_rates),
.list = fsl_sai_rates,
};
@@ -637,6 +637,8 @@
static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
+ case FSL_SAI_TCSR:
+ case FSL_SAI_RCSR:
case FSL_SAI_TFR:
case FSL_SAI_RFR:
case FSL_SAI_TDR:
@@ -681,6 +683,7 @@
.readable_reg = fsl_sai_readable_reg,
.volatile_reg = fsl_sai_volatile_reg,
.writeable_reg = fsl_sai_writeable_reg,
+ .cache_type = REGCACHE_FLAT,
};
static int fsl_sai_probe(struct platform_device *pdev)
@@ -801,11 +804,42 @@
{ .compatible = "fsl,imx6sx-sai", },
{ /* sentinel */ }
};
+MODULE_DEVICE_TABLE(of, fsl_sai_ids);
+
+#ifdef CONFIG_PM_SLEEP
+static int fsl_sai_suspend(struct device *dev)
+{
+ struct fsl_sai *sai = dev_get_drvdata(dev);
+
+ regcache_cache_only(sai->regmap, true);
+ regcache_mark_dirty(sai->regmap);
+
+ return 0;
+}
+
+static int fsl_sai_resume(struct device *dev)
+{
+ struct fsl_sai *sai = dev_get_drvdata(dev);
+
+ regcache_cache_only(sai->regmap, false);
+ regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+ msleep(1);
+ regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+ return regcache_sync(sai->regmap);
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_sai_pm_ops = {
+ SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume)
+};
static struct platform_driver fsl_sai_driver = {
.probe = fsl_sai_probe,
.driver = {
.name = "fsl-sai",
+ .pm = &fsl_sai_pm_ops,
.of_match_table = fsl_sai_ids,
},
};
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index ab729f2..3d59bb6 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -108,6 +108,8 @@
struct clk *sysclk;
struct snd_dmaengine_dai_dma_data dma_params_tx;
struct snd_dmaengine_dai_dma_data dma_params_rx;
+ /* regcache for SRPC */
+ u32 regcache_srpc;
};
/* DPLL locked and lock loss interrupt handler */
@@ -300,6 +302,8 @@
struct regmap *regmap = spdif_priv->regmap;
u32 val, cycle = 1000;
+ regcache_cache_bypass(regmap, true);
+
regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET);
/*
@@ -310,6 +314,10 @@
regmap_read(regmap, REG_SPDIF_SCR, &val);
} while ((val & SCR_SOFT_RESET) && cycle--);
+ regcache_cache_bypass(regmap, false);
+ regcache_mark_dirty(regmap);
+ regcache_sync(regmap);
+
if (cycle)
return 0;
else
@@ -997,6 +1005,14 @@
};
/* FSL SPDIF REGMAP */
+static const struct reg_default fsl_spdif_reg_defaults[] = {
+ {0x0, 0x00000400},
+ {0x4, 0x00000000},
+ {0xc, 0x00000000},
+ {0x34, 0x00000000},
+ {0x38, 0x00000000},
+ {0x50, 0x00020f00},
+};
static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
{
@@ -1022,6 +1038,26 @@
}
}
+static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_SPDIF_SRPC:
+ case REG_SPDIF_SIS:
+ case REG_SPDIF_SRL:
+ case REG_SPDIF_SRR:
+ case REG_SPDIF_SRCSH:
+ case REG_SPDIF_SRCSL:
+ case REG_SPDIF_SRU:
+ case REG_SPDIF_SRQ:
+ case REG_SPDIF_STL:
+ case REG_SPDIF_STR:
+ case REG_SPDIF_SRFM:
+ return true;
+ default:
+ return false;
+ }
+}
+
static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -1047,8 +1083,12 @@
.val_bits = 32,
.max_register = REG_SPDIF_STC,
+ .reg_defaults = fsl_spdif_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(fsl_spdif_reg_defaults),
.readable_reg = fsl_spdif_readable_reg,
+ .volatile_reg = fsl_spdif_volatile_reg,
.writeable_reg = fsl_spdif_writeable_reg,
+ .cache_type = REGCACHE_RBTREE,
};
static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
@@ -1271,6 +1311,38 @@
return ret;
}
+#ifdef CONFIG_PM_SLEEP
+static int fsl_spdif_suspend(struct device *dev)
+{
+ struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev);
+
+ regmap_read(spdif_priv->regmap, REG_SPDIF_SRPC,
+ &spdif_priv->regcache_srpc);
+
+ regcache_cache_only(spdif_priv->regmap, true);
+ regcache_mark_dirty(spdif_priv->regmap);
+
+ return 0;
+}
+
+static int fsl_spdif_resume(struct device *dev)
+{
+ struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev);
+
+ regcache_cache_only(spdif_priv->regmap, false);
+
+ regmap_update_bits(spdif_priv->regmap, REG_SPDIF_SRPC,
+ SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK,
+ spdif_priv->regcache_srpc);
+
+ return regcache_sync(spdif_priv->regmap);
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_spdif_pm = {
+ SET_SYSTEM_SLEEP_PM_OPS(fsl_spdif_suspend, fsl_spdif_resume)
+};
+
static const struct of_device_id fsl_spdif_dt_ids[] = {
{ .compatible = "fsl,imx35-spdif", },
{ .compatible = "fsl,vf610-spdif", },
@@ -1282,6 +1354,7 @@
.driver = {
.name = "fsl-spdif-dai",
.of_match_table = fsl_spdif_dt_ids,
+ .pm = &fsl_spdif_pm,
},
.probe = fsl_spdif_probe,
};
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 37c5cd4..95d2392 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -111,12 +111,75 @@
struct fsl_ssi_reg_val rx;
struct fsl_ssi_reg_val tx;
};
+
+static const struct reg_default fsl_ssi_reg_defaults[] = {
+ {0x10, 0x00000000},
+ {0x18, 0x00003003},
+ {0x1c, 0x00000200},
+ {0x20, 0x00000200},
+ {0x24, 0x00040000},
+ {0x28, 0x00040000},
+ {0x38, 0x00000000},
+ {0x48, 0x00000000},
+ {0x4c, 0x00000000},
+ {0x54, 0x00000000},
+ {0x58, 0x00000000},
+};
+
+static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CCSR_SSI_SACCEN:
+ case CCSR_SSI_SACCDIS:
+ return false;
+ default:
+ return true;
+ }
+}
+
+static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CCSR_SSI_STX0:
+ case CCSR_SSI_STX1:
+ case CCSR_SSI_SRX0:
+ case CCSR_SSI_SRX1:
+ case CCSR_SSI_SISR:
+ case CCSR_SSI_SFCSR:
+ case CCSR_SSI_SACADD:
+ case CCSR_SSI_SACDAT:
+ case CCSR_SSI_SATAG:
+ case CCSR_SSI_SACCST:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_ssi_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CCSR_SSI_SRX0:
+ case CCSR_SSI_SRX1:
+ case CCSR_SSI_SACCST:
+ return false;
+ default:
+ return true;
+ }
+}
+
static const struct regmap_config fsl_ssi_regconfig = {
.max_register = CCSR_SSI_SACCDIS,
.reg_bits = 32,
.val_bits = 32,
.reg_stride = 4,
.val_format_endian = REGMAP_ENDIAN_NATIVE,
+ .reg_defaults = fsl_ssi_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults),
+ .readable_reg = fsl_ssi_readable_reg,
+ .volatile_reg = fsl_ssi_volatile_reg,
+ .writeable_reg = fsl_ssi_writeable_reg,
+ .cache_type = REGCACHE_RBTREE,
};
struct fsl_ssi_soc_data {
@@ -176,6 +239,9 @@
unsigned int baudclk_streams;
unsigned int bitclk_freq;
+ /*regcache for SFCSR*/
+ u32 regcache_sfcsr;
+
/* DMA params */
struct snd_dmaengine_dai_dma_data dma_params_tx;
struct snd_dmaengine_dai_dma_data dma_params_rx;
@@ -1514,10 +1580,46 @@
return 0;
}
+#ifdef CONFIG_PM_SLEEP
+static int fsl_ssi_suspend(struct device *dev)
+{
+ struct fsl_ssi_private *ssi_private = dev_get_drvdata(dev);
+ struct regmap *regs = ssi_private->regs;
+
+ regmap_read(regs, CCSR_SSI_SFCSR,
+ &ssi_private->regcache_sfcsr);
+
+ regcache_cache_only(regs, true);
+ regcache_mark_dirty(regs);
+
+ return 0;
+}
+
+static int fsl_ssi_resume(struct device *dev)
+{
+ struct fsl_ssi_private *ssi_private = dev_get_drvdata(dev);
+ struct regmap *regs = ssi_private->regs;
+
+ regcache_cache_only(regs, false);
+
+ regmap_update_bits(regs, CCSR_SSI_SFCSR,
+ CCSR_SSI_SFCSR_RFWM1_MASK | CCSR_SSI_SFCSR_TFWM1_MASK |
+ CCSR_SSI_SFCSR_RFWM0_MASK | CCSR_SSI_SFCSR_TFWM0_MASK,
+ ssi_private->regcache_sfcsr);
+
+ return regcache_sync(regs);
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_ssi_pm = {
+ SET_SYSTEM_SLEEP_PM_OPS(fsl_ssi_suspend, fsl_ssi_resume)
+};
+
static struct platform_driver fsl_ssi_driver = {
.driver = {
.name = "fsl-ssi-dai",
.of_match_table = fsl_ssi_ids,
+ .pm = &fsl_ssi_pm,
},
.probe = fsl_ssi_probe,
.remove = fsl_ssi_remove,
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 33da26a..a407e83 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -89,6 +89,7 @@
static struct platform_driver imx_spdif_driver = {
.driver = {
.name = "imx-spdif",
+ .pm = &snd_soc_pm_ops,
.of_match_table = imx_spdif_dt_ids,
},
.probe = imx_spdif_audio_probe,
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 3ff76d4..54c3320 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -151,7 +151,9 @@
}
if (set->slots) {
- ret = snd_soc_dai_set_tdm_slot(dai, 0, 0,
+ ret = snd_soc_dai_set_tdm_slot(dai,
+ set->tx_slot_mask,
+ set->rx_slot_mask,
set->slots,
set->slot_width);
if (ret && ret != -ENOTSUPP) {
@@ -243,7 +245,9 @@
return ret;
/* Parse TDM slot */
- ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
+ ret = snd_soc_of_parse_tdm_slot(np, &dai->tx_slot_mask,
+ &dai->rx_slot_mask,
+ &dai->slots, &dai->slot_width);
if (ret)
return ret;
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 05fde5e6e..7b778ab 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -12,6 +12,7 @@
config SND_SST_MFLD_PLATFORM
tristate
+ select SND_SOC_COMPRESS
config SND_SST_IPC
tristate
@@ -138,4 +139,18 @@
config SND_SOC_INTEL_SKYLAKE
tristate
select SND_HDA_EXT_CORE
+ select SND_SOC_TOPOLOGY
select SND_SOC_INTEL_SST
+
+config SND_SOC_INTEL_SKL_RT286_MACH
+ tristate "ASoC Audio driver for SKL with RT286 I2S mode"
+ depends on X86 && ACPI
+ select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SKYLAKE
+ select SND_SOC_RT286
+ select SND_SOC_DMIC
+ help
+ This adds support for ASoC machine driver for Skylake platforms
+ with RT286 I2S audio codec.
+ Say Y if you have such a device
+ If unsure select "N".
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 683e501..0487cfa 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -368,23 +368,6 @@
kfree(stream);
}
-static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai,
- struct snd_pcm_substream *substream)
-{
- struct sst_data *sst = snd_soc_dai_get_drvdata(dai);
- struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
- struct sst_runtime_stream *stream =
- substream->runtime->private_data;
- u32 str_id = stream->stream_info.str_id;
- unsigned int pipe_id;
-
- pipe_id = map[str_id].device_id;
-
- dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n",
- pipe_id, str_id);
- return pipe_id;
-}
-
static int sst_media_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -529,7 +512,7 @@
},
{
.name = "compress-cpu-dai",
- .compress_dai = 1,
+ .compress_new = snd_soc_new_compress,
.ops = &sst_compr_dai_ops,
.playback = {
.stream_name = "Compress Playback",
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index cb94895..371c456 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -6,6 +6,7 @@
snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o
+snd-soc-skl_rt286-objs := skl_rt286.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
@@ -15,3 +16,4 @@
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o
+obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 8bafaf6..3f8a1e1 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -266,18 +266,11 @@
{
broadwell_rt286.dev = &pdev->dev;
- return snd_soc_register_card(&broadwell_rt286);
-}
-
-static int broadwell_audio_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&broadwell_rt286);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286);
}
static struct platform_driver broadwell_audio = {
.probe = broadwell_audio_probe,
- .remove = broadwell_audio_remove,
.driver = {
.name = "broadwell-audio",
},
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index c445312..7a5c9a3 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -117,20 +117,10 @@
return 0;
}
-static unsigned int rates_48000[] = {
- 48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
- .count = ARRAY_SIZE(rates_48000),
- .list = rates_48000,
-};
-
static int byt_aif1_startup(struct snd_pcm_substream *substream)
{
- return snd_pcm_hw_constraint_list(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_48000);
+ return snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, 48000);
}
static struct snd_soc_ops byt_aif1_ops = {
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 49f4869..4e2fcf1 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -193,20 +193,10 @@
return 0;
}
-static unsigned int rates_48000[] = {
- 48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
- .count = ARRAY_SIZE(rates_48000),
- .list = rates_48000,
-};
-
static int cht_aif1_startup(struct snd_pcm_substream *substream)
{
- return snd_pcm_hw_constraint_list(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_48000);
+ return snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, 48000);
}
static int cht_max98090_headset_init(struct snd_soc_component *component)
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 7be8461..38d65a3 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -235,20 +235,10 @@
return 0;
}
-static unsigned int rates_48000[] = {
- 48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
- .count = ARRAY_SIZE(rates_48000),
- .list = rates_48000,
-};
-
static int cht_aif1_startup(struct snd_pcm_substream *substream)
{
- return snd_pcm_hw_constraint_list(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_48000);
+ return snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, 48000);
}
static struct snd_soc_ops cht_aif1_ops = {
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 23fe040..5621ccd 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -222,20 +222,10 @@
return 0;
}
-static unsigned int rates_48000[] = {
- 48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
- .count = ARRAY_SIZE(rates_48000),
- .list = rates_48000,
-};
-
static int cht_aif1_startup(struct snd_pcm_substream *substream)
{
- return snd_pcm_hw_constraint_list(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_48000);
+ return snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, 48000);
}
static struct snd_soc_ops cht_aif1_ops = {
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
new file mode 100644
index 0000000..a73a431
--- /dev/null
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -0,0 +1,259 @@
+/*
+ * Intel Skylake I2S Machine Driver
+ *
+ * Copyright (C) 2014-2015, Intel Corporation. All rights reserved.
+ *
+ * Modified from:
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+#include "../../codecs/rt286.h"
+
+static struct snd_soc_jack skylake_headset;
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin skylake_headset_pins[] = {
+ {
+ .pin = "Mic Jack",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+static const struct snd_kcontrol_new skylake_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Mic Jack"),
+};
+
+static const struct snd_soc_dapm_widget skylake_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("DMIC2", NULL),
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_route skylake_rt286_map[] = {
+ /* speaker */
+ {"Speaker", NULL, "SPOR"},
+ {"Speaker", NULL, "SPOL"},
+
+ /* HP jack connectors - unknown if we have jack deteck */
+ {"Headphone Jack", NULL, "HPO Pin"},
+
+ /* other jacks */
+ {"MIC1", NULL, "Mic Jack"},
+
+ /* digital mics */
+ {"DMIC1 Pin", NULL, "DMIC2"},
+ {"DMIC AIF", NULL, "SoC DMIC"},
+
+ /* CODEC BE connections */
+ { "AIF1 Playback", NULL, "ssp0 Tx"},
+ { "ssp0 Tx", NULL, "codec0_out"},
+ { "ssp0 Tx", NULL, "codec1_out"},
+
+ { "codec0_in", NULL, "ssp0 Rx" },
+ { "codec1_in", NULL, "ssp0 Rx" },
+ { "ssp0 Rx", NULL, "AIF1 Capture" },
+
+ { "dmic01_hifi", NULL, "DMIC01 Rx" },
+ { "DMIC01 Rx", NULL, "Capture" },
+
+ { "hif1", NULL, "iDisp Tx"},
+ { "iDisp Tx", NULL, "iDisp_out"},
+
+};
+
+static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ int ret;
+
+ ret = snd_soc_card_jack_new(rtd->card, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0,
+ &skylake_headset,
+ skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins));
+
+ if (ret)
+ return ret;
+
+ rt286_mic_detect(codec, &skylake_headset);
+
+ return 0;
+}
+
+
+static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The output is 48KHz, stereo, 16bits */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
+
+ return 0;
+}
+
+static int skylake_rt286_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret);
+
+ return ret;
+}
+
+static struct snd_soc_ops skylake_rt286_ops = {
+ .hw_params = skylake_rt286_hw_params,
+};
+
+/* skylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link skylake_rt286_dais[] = {
+ /* Front End DAI links */
+ {
+ .name = "Skl Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:1f.3",
+ .nonatomic = 1,
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST,
+ SND_SOC_DPCM_TRIGGER_POST
+ },
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Skl Audio Capture Port",
+ .stream_name = "Audio Record",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:1f.3",
+ .nonatomic = 1,
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {
+ SND_SOC_DPCM_TRIGGER_POST,
+ SND_SOC_DPCM_TRIGGER_POST
+ },
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "Skl Audio Reference cap",
+ .stream_name = "refcap",
+ .cpu_dai_name = "Reference Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:1f.3",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .ignore_suspend = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "SSP0-Codec",
+ .be_id = 0,
+ .cpu_dai_name = "SSP0 Pin",
+ .platform_name = "0000:00:1f.3",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT343A:00",
+ .codec_dai_name = "rt286-aif1",
+ .init = skylake_rt286_codec_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = skylake_ssp0_fixup,
+ .ops = &skylake_rt286_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "dmic01",
+ .be_id = 1,
+ .cpu_dai_name = "DMIC01 Pin",
+ .codec_name = "dmic-codec",
+ .codec_dai_name = "dmic-hifi",
+ .platform_name = "0000:00:1f.3",
+ .ignore_suspend = 1,
+ .dpcm_capture = 1,
+ .no_pcm = 1,
+ },
+};
+
+/* skylake audio machine driver for SPT + RT286S */
+static struct snd_soc_card skylake_rt286 = {
+ .name = "skylake-rt286",
+ .owner = THIS_MODULE,
+ .dai_link = skylake_rt286_dais,
+ .num_links = ARRAY_SIZE(skylake_rt286_dais),
+ .controls = skylake_controls,
+ .num_controls = ARRAY_SIZE(skylake_controls),
+ .dapm_widgets = skylake_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(skylake_widgets),
+ .dapm_routes = skylake_rt286_map,
+ .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map),
+ .fully_routed = true,
+};
+
+static int skylake_audio_probe(struct platform_device *pdev)
+{
+ skylake_rt286.dev = &pdev->dev;
+
+ return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286);
+}
+
+static struct platform_driver skylake_audio = {
+ .probe = skylake_audio_probe,
+ .driver = {
+ .name = "skl_alc286s_i2s",
+ },
+};
+
+module_platform_driver(skylake_audio)
+
+/* Module information */
+MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah@intel.com>");
+MODULE_DESCRIPTION("Intel SST Audio for Skylake");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:skl_alc286s_i2s");
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index f24154c..d910558 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -1,7 +1,11 @@
-snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o
+snd-soc-sst-dsp-objs := sst-dsp.o
snd-soc-sst-acpi-objs := sst-acpi.o
snd-soc-sst-ipc-objs := sst-ipc.o
+ifneq ($(CONFIG_DW_DMAC_CORE),)
+snd-soc-sst-dsp-objs += sst-firmware.o
+endif
+
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h
index cbd568e..2151652 100644
--- a/sound/soc/intel/common/sst-dsp-priv.h
+++ b/sound/soc/intel/common/sst-dsp-priv.h
@@ -314,6 +314,7 @@
int sst_state;
struct skl_cl_dev cl_dev;
u32 intr_status;
+ const struct firmware *fw;
};
/* Size optimised DRAM/IRAM memcpy */
diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c
index a627236..c9452e0 100644
--- a/sound/soc/intel/common/sst-dsp.c
+++ b/sound/soc/intel/common/sst-dsp.c
@@ -420,6 +420,7 @@
}
EXPORT_SYMBOL_GPL(sst_dsp_inbox_read);
+#if IS_ENABLED(CONFIG_DW_DMAC_CORE)
struct sst_dsp *sst_dsp_new(struct device *dev,
struct sst_dsp_device *sst_dev, struct sst_pdata *pdata)
{
@@ -484,6 +485,7 @@
sst_dma_free(sst->dma);
}
EXPORT_SYMBOL_GPL(sst_dsp_free);
+#endif
/* Module information */
MODULE_AUTHOR("Liam Girdwood");
diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h
index 1f45f18..859f0de 100644
--- a/sound/soc/intel/common/sst-dsp.h
+++ b/sound/soc/intel/common/sst-dsp.h
@@ -216,10 +216,12 @@
void *dsp;
};
+#if IS_ENABLED(CONFIG_DW_DMAC_CORE)
/* Initialization */
struct sst_dsp *sst_dsp_new(struct device *dev,
struct sst_dsp_device *sst_dev, struct sst_pdata *pdata);
void sst_dsp_free(struct sst_dsp *sst);
+#endif
/* SHIM Read / Write */
void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index ebcca6d..1636a1e 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -26,7 +26,6 @@
#include <linux/acpi.h>
/* supported DMA engine drivers */
-#include <linux/platform_data/dma-dw.h>
#include <linux/dma/dw.h>
#include <asm/page.h>
@@ -169,12 +168,6 @@
return ret;
}
-static struct dw_dma_platform_data dw_pdata = {
- .is_private = 1,
- .chan_allocation_order = CHAN_ALLOCATION_ASCENDING,
- .chan_priority = CHAN_PRIORITY_ASCENDING,
-};
-
static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem,
int irq)
{
@@ -195,7 +188,8 @@
return ERR_PTR(err);
chip->dev = dev;
- err = dw_dma_probe(chip, &dw_pdata);
+
+ err = dw_dma_probe(chip, NULL);
if (err)
return ERR_PTR(err);
diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index 27db221..914b6da 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -1,4 +1,5 @@
-snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o
+snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o \
+skl-topology.o
obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 826d4fd..50a1095 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -54,6 +54,24 @@
return 0;
}
+#define NOTIFICATION_PARAM_ID 3
+#define NOTIFICATION_MASK 0xf
+
+/* disable notfication for underruns/overruns from firmware module */
+static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
+{
+ struct notification_mask mask;
+ struct skl_ipc_large_config_msg msg = {0};
+
+ mask.notify = NOTIFICATION_MASK;
+ mask.enable = enable;
+
+ msg.large_param_id = NOTIFICATION_PARAM_ID;
+ msg.param_data_size = sizeof(mask);
+
+ skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask);
+}
+
int skl_init_dsp(struct skl *skl)
{
void __iomem *mmio_base;
@@ -79,7 +97,10 @@
ret = skl_sst_dsp_init(bus->dev, mmio_base, irq,
loader_ops, &skl->skl_sst);
+ if (ret < 0)
+ return ret;
+ skl_dsp_enable_notification(skl->skl_sst, false);
dev_dbg(bus->dev, "dsp registration status=%d\n", ret);
return ret;
@@ -122,6 +143,7 @@
int skl_resume_dsp(struct skl *skl)
{
struct skl_sst *ctx = skl->skl_sst;
+ int ret;
/* if ppcap is not supported return 0 */
if (!skl->ebus.ppcap)
@@ -131,7 +153,12 @@
snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true);
snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true);
- return skl_dsp_wake(ctx->dsp);
+ ret = skl_dsp_wake(ctx->dsp);
+ if (ret < 0)
+ return ret;
+
+ skl_dsp_enable_notification(skl->skl_sst, false);
+ return ret;
}
enum skl_bitdepth skl_get_bit_depth(int params)
@@ -294,6 +321,7 @@
(mconfig->formats_config.caps_size) / 4;
}
+#define SKL_NON_GATEWAY_CPR_NODE_ID 0xFFFFFFFF
/*
* Calculate the gatewat settings required for copier module, type of
* gateway and index of gateway to use
@@ -303,6 +331,7 @@
struct skl_cpr_cfg *cpr_mconfig)
{
union skl_connector_node_id node_id = {0};
+ union skl_ssp_dma_node ssp_node = {0};
struct skl_pipe_params *params = mconfig->pipe->p_params;
switch (mconfig->dev_type) {
@@ -320,9 +349,9 @@
(SKL_CONN_SOURCE == mconfig->hw_conn_type) ?
SKL_DMA_I2S_LINK_OUTPUT_CLASS :
SKL_DMA_I2S_LINK_INPUT_CLASS;
- node_id.node.vindex = params->host_dma_id +
- (mconfig->time_slot << 1) +
- (mconfig->vbus_id << 3);
+ ssp_node.dma_node.time_slot_index = mconfig->time_slot;
+ ssp_node.dma_node.i2s_instance = mconfig->vbus_id;
+ node_id.node.vindex = ssp_node.val;
break;
case SKL_DEVICE_DMIC:
@@ -339,13 +368,18 @@
node_id.node.vindex = params->link_dma_id;
break;
- default:
+ case SKL_DEVICE_HDAHOST:
node_id.node.dma_type =
(SKL_CONN_SOURCE == mconfig->hw_conn_type) ?
SKL_DMA_HDA_HOST_OUTPUT_CLASS :
SKL_DMA_HDA_HOST_INPUT_CLASS;
node_id.node.vindex = params->host_dma_id;
break;
+
+ default:
+ cpr_mconfig->gtw_cfg.node_id = SKL_NON_GATEWAY_CPR_NODE_ID;
+ cpr_mconfig->cpr_feature_mask = 0;
+ return;
}
cpr_mconfig->gtw_cfg.node_id = node_id.val;
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 13036b1..b0c7bd1 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -25,7 +25,7 @@
#define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS"
-void __iomem *skl_nhlt_init(struct device *dev)
+void *skl_nhlt_init(struct device *dev)
{
acpi_handle handle;
union acpi_object *obj;
@@ -40,17 +40,17 @@
if (obj && obj->type == ACPI_TYPE_BUFFER) {
nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer;
- return ioremap_cache(nhlt_ptr->min_addr, nhlt_ptr->length);
+ return memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
+ MEMREMAP_WB);
}
dev_err(dev, "device specific method to extract NHLT blob failed\n");
return NULL;
}
-void skl_nhlt_free(void __iomem *addr)
+void skl_nhlt_free(void *addr)
{
- iounmap(addr);
- addr = NULL;
+ memunmap(addr);
}
static struct nhlt_specific_cfg *skl_get_specific_cfg(
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 7d617bf..a2f94ce 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -24,6 +24,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "skl.h"
+#include "skl-topology.h"
#define HDA_MONO 1
#define HDA_STEREO 2
@@ -115,7 +116,7 @@
dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
ret = pm_runtime_get_sync(dai->dev);
- if (ret)
+ if (ret < 0)
return ret;
stream = snd_hdac_ext_stream_assign(ebus, substream,
@@ -214,6 +215,8 @@
struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ struct skl_pipe_params p_params = {0};
+ struct skl_module_cfg *m_cfg;
int ret, dma_id;
dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
@@ -228,6 +231,16 @@
dma_id = hdac_stream(stream)->stream_tag - 1;
dev_dbg(dai->dev, "dma_id=%d\n", dma_id);
+ p_params.s_fmt = snd_pcm_format_width(params_format(params));
+ p_params.ch = params_channels(params);
+ p_params.s_freq = params_rate(params);
+ p_params.host_dma_id = dma_id;
+ p_params.stream = substream->stream;
+
+ m_cfg = skl_tplg_fe_get_cpr_module(dai, p_params.stream);
+ if (m_cfg)
+ skl_tplg_update_pipe_params(dai->dev, m_cfg, &p_params);
+
return 0;
}
@@ -268,6 +281,46 @@
return skl_substream_free_pages(ebus_to_hbus(ebus), substream);
}
+static int skl_be_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct skl_pipe_params p_params = {0};
+
+ p_params.s_fmt = snd_pcm_format_width(params_format(params));
+ p_params.ch = params_channels(params);
+ p_params.s_freq = params_rate(params);
+ p_params.stream = substream->stream;
+ skl_tplg_be_update_params(dai, &p_params);
+
+ return 0;
+}
+
+static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct skl *skl = get_skl_ctx(dai->dev);
+ struct skl_sst *ctx = skl->skl_sst;
+ struct skl_module_cfg *mconfig;
+
+ mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
+ if (!mconfig)
+ return -EIO;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ return skl_run_pipe(ctx, mconfig->pipe);
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ return skl_stop_pipe(ctx, mconfig->pipe);
+
+ default:
+ return 0;
+ }
+}
+
static int skl_link_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -277,9 +330,8 @@
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
struct skl_dma_params *dma_params;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int dma_id;
+ struct skl_pipe_params p_params = {0};
- pr_debug("%s\n", __func__);
link_dev = snd_hdac_ext_stream_assign(ebus, substream,
HDAC_EXT_STREAM_TYPE_LINK);
if (!link_dev)
@@ -293,7 +345,14 @@
if (dma_params)
dma_params->stream_tag = hdac_stream(link_dev)->stream_tag;
snd_soc_dai_set_dma_data(codec_dai, substream, (void *)dma_params);
- dma_id = hdac_stream(link_dev)->stream_tag - 1;
+
+ p_params.s_fmt = snd_pcm_format_width(params_format(params));
+ p_params.ch = params_channels(params);
+ p_params.s_freq = params_rate(params);
+ p_params.stream = substream->stream;
+ p_params.link_dma_id = hdac_stream(link_dev)->stream_tag - 1;
+
+ skl_tplg_be_update_params(dai, &p_params);
return 0;
}
@@ -308,27 +367,12 @@
unsigned int format_val = 0;
struct skl_dma_params *dma_params;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_pcm_hw_params *params;
- struct snd_interval *channels, *rate;
struct hdac_ext_link *link;
- dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
if (link_dev->link_prepared) {
dev_dbg(dai->dev, "already stream is prepared - returning\n");
return 0;
}
- params = devm_kzalloc(dai->dev, sizeof(*params), GFP_KERNEL);
- if (params == NULL)
- return -ENOMEM;
-
- channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
- channels->min = channels->max = substream->runtime->channels;
- rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
- rate->min = rate->max = substream->runtime->rate;
- snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
- SNDRV_PCM_HW_PARAM_FIRST_MASK],
- substream->runtime->format);
-
dma_params = (struct skl_dma_params *)
snd_soc_dai_get_dma_data(codec_dai, substream);
@@ -399,13 +443,13 @@
return 0;
}
-static int skl_hda_be_startup(struct snd_pcm_substream *substream,
+static int skl_be_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
return pm_runtime_get_sync(dai->dev);
}
-static void skl_hda_be_shutdown(struct snd_pcm_substream *substream,
+static void skl_be_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
pm_runtime_mark_last_busy(dai->dev);
@@ -418,20 +462,28 @@
.prepare = skl_pcm_prepare,
.hw_params = skl_pcm_hw_params,
.hw_free = skl_pcm_hw_free,
+ .trigger = skl_pcm_trigger,
};
static struct snd_soc_dai_ops skl_dmic_dai_ops = {
- .startup = skl_hda_be_startup,
- .shutdown = skl_hda_be_shutdown,
+ .startup = skl_be_startup,
+ .hw_params = skl_be_hw_params,
+ .shutdown = skl_be_shutdown,
+};
+
+static struct snd_soc_dai_ops skl_be_ssp_dai_ops = {
+ .startup = skl_be_startup,
+ .hw_params = skl_be_hw_params,
+ .shutdown = skl_be_shutdown,
};
static struct snd_soc_dai_ops skl_link_dai_ops = {
- .startup = skl_hda_be_startup,
+ .startup = skl_be_startup,
.prepare = skl_link_pcm_prepare,
.hw_params = skl_link_hw_params,
.hw_free = skl_link_hw_free,
.trigger = skl_link_pcm_trigger,
- .shutdown = skl_hda_be_shutdown,
+ .shutdown = skl_be_shutdown,
};
static struct snd_soc_dai_driver skl_platform_dai[] = {
@@ -488,6 +540,24 @@
},
/* BE CPU Dais */
{
+ .name = "SSP0 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp0 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp0 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
.name = "iDisp Pin",
.ops = &skl_link_dai_ops,
.playback = {
@@ -510,17 +580,6 @@
},
},
{
- .name = "DMIC23 Pin",
- .ops = &skl_dmic_dai_ops,
- .capture = {
- .stream_name = "DMIC23 Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
-},
-{
.name = "HD-Codec Pin",
.ops = &skl_link_dai_ops,
.playback = {
@@ -538,28 +597,6 @@
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
-{
- .name = "HD-Codec-SPK Pin",
- .ops = &skl_link_dai_ops,
- .playback = {
- .stream_name = "HD-Codec-SPK Tx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
-{
- .name = "HD-Codec-AMIC Pin",
- .ops = &skl_link_dai_ops,
- .capture = {
- .stream_name = "HD-Codec-AMIC Rx",
- .channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
-},
};
static int skl_platform_open(struct snd_pcm_substream *substream)
@@ -577,7 +614,7 @@
return 0;
}
-static int skl_pcm_trigger(struct snd_pcm_substream *substream,
+static int skl_coupled_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct hdac_ext_bus *ebus = get_bus_ctx(substream);
@@ -651,7 +688,7 @@
return 0;
}
-static int skl_dsp_trigger(struct snd_pcm_substream *substream,
+static int skl_decoupled_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct hdac_ext_bus *ebus = get_bus_ctx(substream);
@@ -708,9 +745,9 @@
struct hdac_ext_bus *ebus = get_bus_ctx(substream);
if (ebus->ppcap)
- return skl_dsp_trigger(substream, cmd);
+ return skl_decoupled_trigger(substream, cmd);
else
- return skl_pcm_trigger(substream, cmd);
+ return skl_coupled_trigger(substream, cmd);
}
/* calculate runtime delay from LPIB */
@@ -877,7 +914,17 @@
return retval;
}
+static int skl_platform_soc_probe(struct snd_soc_platform *platform)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(platform->dev);
+
+ if (ebus->ppcap)
+ return skl_tplg_init(platform, ebus);
+
+ return 0;
+}
static struct snd_soc_platform_driver skl_platform_drv = {
+ .probe = skl_platform_soc_probe,
.ops = &skl_platform_ops,
.pcm_new = skl_pcm_new,
.pcm_free = skl_pcm_free,
@@ -890,6 +937,11 @@
int skl_platform_register(struct device *dev)
{
int ret;
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct skl *skl = ebus_to_skl(ebus);
+
+ INIT_LIST_HEAD(&skl->ppl_list);
+ INIT_LIST_HEAD(&skl->dapm_path_list);
ret = snd_soc_register_platform(dev, &skl_platform_drv);
if (ret) {
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index 94875b0..1bfb7f6 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -175,7 +175,7 @@
/* poll with timeout to check if operation successful */
return sst_dsp_register_poll(ctx,
SKL_ADSP_REG_ADSPCS,
- SKL_ADSPCS_SPA_MASK,
+ SKL_ADSPCS_CPA_MASK,
0,
SKL_DSP_PD_TO,
"Power down");
@@ -262,6 +262,11 @@
val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPIS);
ctx->intr_status = val;
+ if (val == 0xffffffff) {
+ spin_unlock(&ctx->spinlock);
+ return IRQ_NONE;
+ }
+
if (val & SKL_ADSPIS_IPC) {
skl_ipc_int_disable(ctx);
result = IRQ_WAKE_THREAD;
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index 937a0a3..3345ea0 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -464,6 +464,18 @@
SKL_ADSP_REG_HIPCCTL_BUSY, SKL_ADSP_REG_HIPCCTL_BUSY);
}
+void skl_ipc_op_int_disable(struct sst_dsp *ctx)
+{
+ /* disable IPC DONE interrupt */
+ sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL,
+ SKL_ADSP_REG_HIPCCTL_DONE, 0);
+
+ /* Disable IPC BUSY interrupt */
+ sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL,
+ SKL_ADSP_REG_HIPCCTL_BUSY, 0);
+
+}
+
bool skl_ipc_int_status(struct sst_dsp *ctx)
{
return sst_dsp_shim_read_unlocked(ctx,
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h
index 9f5f672..f1a154e 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.h
+++ b/sound/soc/intel/skylake/skl-sst-ipc.h
@@ -116,6 +116,7 @@
void skl_ipc_int_enable(struct sst_dsp *dsp);
void skl_ipc_op_int_enable(struct sst_dsp *ctx);
+void skl_ipc_op_int_disable(struct sst_dsp *ctx);
void skl_ipc_int_disable(struct sst_dsp *dsp);
bool skl_ipc_int_status(struct sst_dsp *dsp);
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index c18ea51..3b83dc9 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -70,15 +70,31 @@
static int skl_load_base_firmware(struct sst_dsp *ctx)
{
int ret = 0, i;
- const struct firmware *fw = NULL;
struct skl_sst *skl = ctx->thread_context;
u32 reg;
- ret = request_firmware(&fw, "dsp_fw_release.bin", ctx->dev);
+ skl->boot_complete = false;
+ init_waitqueue_head(&skl->boot_wait);
+
+ if (ctx->fw == NULL) {
+ ret = request_firmware(&ctx->fw, "dsp_fw_release.bin", ctx->dev);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Request firmware failed %d\n", ret);
+ skl_dsp_disable_core(ctx);
+ return -EIO;
+ }
+ }
+
+ ret = skl_dsp_boot(ctx);
if (ret < 0) {
- dev_err(ctx->dev, "Request firmware failed %d\n", ret);
- skl_dsp_disable_core(ctx);
- return -EIO;
+ dev_err(ctx->dev, "Boot dsp core failed ret: %d", ret);
+ goto skl_load_base_firmware_failed;
+ }
+
+ ret = skl_cldma_prepare(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "CL dma prepare failed : %d", ret);
+ goto skl_load_base_firmware_failed;
}
/* enable Interrupt */
@@ -102,7 +118,7 @@
goto skl_load_base_firmware_failed;
}
- ret = skl_transfer_firmware(ctx, fw->data, fw->size);
+ ret = skl_transfer_firmware(ctx, ctx->fw->data, ctx->fw->size);
if (ret < 0) {
dev_err(ctx->dev, "Transfer firmware failed%d\n", ret);
goto skl_load_base_firmware_failed;
@@ -118,13 +134,12 @@
dev_dbg(ctx->dev, "Download firmware successful%d\n", ret);
skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
}
- release_firmware(fw);
-
return 0;
skl_load_base_firmware_failed:
skl_dsp_disable_core(ctx);
- release_firmware(fw);
+ release_firmware(ctx->fw);
+ ctx->fw = NULL;
return ret;
}
@@ -172,6 +187,12 @@
}
skl_dsp_set_state_locked(ctx, SKL_DSP_RESET);
+ /* disable Interrupt */
+ ctx->cl_dev.ops.cl_cleanup_controller(ctx);
+ skl_cldma_int_disable(ctx);
+ skl_ipc_op_int_disable(ctx);
+ skl_ipc_int_disable(ctx);
+
return ret;
}
@@ -235,22 +256,6 @@
if (ret)
return ret;
- skl->boot_complete = false;
- init_waitqueue_head(&skl->boot_wait);
-
- ret = skl_dsp_boot(sst);
- if (ret < 0) {
- dev_err(skl->dev, "Boot dsp core failed ret: %d", ret);
- goto free_ipc;
- }
-
- ret = skl_cldma_prepare(sst);
- if (ret < 0) {
- dev_err(dev, "CL dma prepare failed : %d", ret);
- goto free_ipc;
- }
-
-
ret = sst->fw_ops.load_fw(sst);
if (ret < 0) {
dev_err(dev, "Load base fw failed : %d", ret);
@@ -262,7 +267,6 @@
return 0;
-free_ipc:
skl_ipc_free(&skl->ipc);
return ret;
}
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
new file mode 100644
index 0000000..a7854c8
--- /dev/null
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -0,0 +1,1252 @@
+/*
+ * skl-topology.c - Implements Platform component ALSA controls/widget
+ * handlers.
+ *
+ * Copyright (C) 2014-2015 Intel Corp
+ * Author: Jeeja KP <jeeja.kp@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.
+ */
+
+#include <linux/slab.h>
+#include <linux/types.h>
+#include <linux/firmware.h>
+#include <sound/soc.h>
+#include <sound/soc-topology.h>
+#include "skl-sst-dsp.h"
+#include "skl-sst-ipc.h"
+#include "skl-topology.h"
+#include "skl.h"
+#include "skl-tplg-interface.h"
+
+#define SKL_CH_FIXUP_MASK (1 << 0)
+#define SKL_RATE_FIXUP_MASK (1 << 1)
+#define SKL_FMT_FIXUP_MASK (1 << 2)
+
+/*
+ * SKL DSP driver modelling uses only few DAPM widgets so for rest we will
+ * ignore. This helpers checks if the SKL driver handles this widget type
+ */
+static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w)
+{
+ switch (w->id) {
+ case snd_soc_dapm_dai_link:
+ case snd_soc_dapm_dai_in:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
+ case snd_soc_dapm_dai_out:
+ case snd_soc_dapm_switch:
+ return false;
+ default:
+ return true;
+ }
+}
+
+/*
+ * Each pipelines needs memory to be allocated. Check if we have free memory
+ * from available pool. Then only add this to pool
+ * This is freed when pipe is deleted
+ * Note: DSP does actual memory management we only keep track for complete
+ * pool
+ */
+static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
+ struct skl_module_cfg *mconfig)
+{
+ struct skl_sst *ctx = skl->skl_sst;
+
+ if (skl->resource.mem + mconfig->pipe->memory_pages >
+ skl->resource.max_mem) {
+ dev_err(ctx->dev,
+ "%s: module_id %d instance %d\n", __func__,
+ mconfig->id.module_id,
+ mconfig->id.instance_id);
+ dev_err(ctx->dev,
+ "exceeds ppl memory available %d mem %d\n",
+ skl->resource.max_mem, skl->resource.mem);
+ return false;
+ }
+
+ skl->resource.mem += mconfig->pipe->memory_pages;
+ return true;
+}
+
+/*
+ * Pipeline needs needs DSP CPU resources for computation, this is
+ * quantified in MCPS (Million Clocks Per Second) required for module/pipe
+ *
+ * Each pipelines needs mcps to be allocated. Check if we have mcps for this
+ * pipe. This adds the mcps to driver counter
+ * This is removed on pipeline delete
+ */
+static bool skl_tplg_alloc_pipe_mcps(struct skl *skl,
+ struct skl_module_cfg *mconfig)
+{
+ struct skl_sst *ctx = skl->skl_sst;
+
+ if (skl->resource.mcps + mconfig->mcps > skl->resource.max_mcps) {
+ dev_err(ctx->dev,
+ "%s: module_id %d instance %d\n", __func__,
+ mconfig->id.module_id, mconfig->id.instance_id);
+ dev_err(ctx->dev,
+ "exceeds ppl memory available %d > mem %d\n",
+ skl->resource.max_mcps, skl->resource.mcps);
+ return false;
+ }
+
+ skl->resource.mcps += mconfig->mcps;
+ return true;
+}
+
+/*
+ * Free the mcps when tearing down
+ */
+static void
+skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig)
+{
+ skl->resource.mcps -= mconfig->mcps;
+}
+
+/*
+ * Free the memory when tearing down
+ */
+static void
+skl_tplg_free_pipe_mem(struct skl *skl, struct skl_module_cfg *mconfig)
+{
+ skl->resource.mem -= mconfig->pipe->memory_pages;
+}
+
+
+static void skl_dump_mconfig(struct skl_sst *ctx,
+ struct skl_module_cfg *mcfg)
+{
+ dev_dbg(ctx->dev, "Dumping config\n");
+ dev_dbg(ctx->dev, "Input Format:\n");
+ dev_dbg(ctx->dev, "channels = %d\n", mcfg->in_fmt.channels);
+ dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->in_fmt.s_freq);
+ dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->in_fmt.ch_cfg);
+ dev_dbg(ctx->dev, "valid bit depth = %d\n",
+ mcfg->in_fmt.valid_bit_depth);
+ dev_dbg(ctx->dev, "Output Format:\n");
+ dev_dbg(ctx->dev, "channels = %d\n", mcfg->out_fmt.channels);
+ dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->out_fmt.s_freq);
+ dev_dbg(ctx->dev, "valid bit depth = %d\n",
+ mcfg->out_fmt.valid_bit_depth);
+ dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt.ch_cfg);
+}
+
+static void skl_tplg_update_params(struct skl_module_fmt *fmt,
+ struct skl_pipe_params *params, int fixup)
+{
+ if (fixup & SKL_RATE_FIXUP_MASK)
+ fmt->s_freq = params->s_freq;
+ if (fixup & SKL_CH_FIXUP_MASK)
+ fmt->channels = params->ch;
+ if (fixup & SKL_FMT_FIXUP_MASK)
+ fmt->valid_bit_depth = params->s_fmt;
+}
+
+/*
+ * A pipeline may have modules which impact the pcm parameters, like SRC,
+ * channel converter, format converter.
+ * We need to calculate the output params by applying the 'fixup'
+ * Topology will tell driver which type of fixup is to be applied by
+ * supplying the fixup mask, so based on that we calculate the output
+ *
+ * Now In FE the pcm hw_params is source/target format. Same is applicable
+ * for BE with its hw_params invoked.
+ * here based on FE, BE pipeline and direction we calculate the input and
+ * outfix and then apply that for a module
+ */
+static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg,
+ struct skl_pipe_params *params, bool is_fe)
+{
+ int in_fixup, out_fixup;
+ struct skl_module_fmt *in_fmt, *out_fmt;
+
+ in_fmt = &m_cfg->in_fmt;
+ out_fmt = &m_cfg->out_fmt;
+
+ if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (is_fe) {
+ in_fixup = m_cfg->params_fixup;
+ out_fixup = (~m_cfg->converter) &
+ m_cfg->params_fixup;
+ } else {
+ out_fixup = m_cfg->params_fixup;
+ in_fixup = (~m_cfg->converter) &
+ m_cfg->params_fixup;
+ }
+ } else {
+ if (is_fe) {
+ out_fixup = m_cfg->params_fixup;
+ in_fixup = (~m_cfg->converter) &
+ m_cfg->params_fixup;
+ } else {
+ in_fixup = m_cfg->params_fixup;
+ out_fixup = (~m_cfg->converter) &
+ m_cfg->params_fixup;
+ }
+ }
+
+ skl_tplg_update_params(in_fmt, params, in_fixup);
+ skl_tplg_update_params(out_fmt, params, out_fixup);
+}
+
+/*
+ * A module needs input and output buffers, which are dependent upon pcm
+ * params, so once we have calculate params, we need buffer calculation as
+ * well.
+ */
+static void skl_tplg_update_buffer_size(struct skl_sst *ctx,
+ struct skl_module_cfg *mcfg)
+{
+ int multiplier = 1;
+
+ if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT)
+ multiplier = 5;
+
+ mcfg->ibs = (mcfg->in_fmt.s_freq / 1000) *
+ (mcfg->in_fmt.channels) *
+ (mcfg->in_fmt.bit_depth >> 3) *
+ multiplier;
+
+ mcfg->obs = (mcfg->out_fmt.s_freq / 1000) *
+ (mcfg->out_fmt.channels) *
+ (mcfg->out_fmt.bit_depth >> 3) *
+ multiplier;
+}
+
+static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w,
+ struct skl_sst *ctx)
+{
+ struct skl_module_cfg *m_cfg = w->priv;
+ struct skl_pipe_params *params = m_cfg->pipe->p_params;
+ int p_conn_type = m_cfg->pipe->conn_type;
+ bool is_fe;
+
+ if (!m_cfg->params_fixup)
+ return;
+
+ dev_dbg(ctx->dev, "Mconfig for widget=%s BEFORE updation\n",
+ w->name);
+
+ skl_dump_mconfig(ctx, m_cfg);
+
+ if (p_conn_type == SKL_PIPE_CONN_TYPE_FE)
+ is_fe = true;
+ else
+ is_fe = false;
+
+ skl_tplg_update_params_fixup(m_cfg, params, is_fe);
+ skl_tplg_update_buffer_size(ctx, m_cfg);
+
+ dev_dbg(ctx->dev, "Mconfig for widget=%s AFTER updation\n",
+ w->name);
+
+ skl_dump_mconfig(ctx, m_cfg);
+}
+
+/*
+ * A pipe can have multiple modules, each of them will be a DAPM widget as
+ * well. While managing a pipeline we need to get the list of all the
+ * widgets in a pipelines, so this helper - skl_tplg_get_pipe_widget() helps
+ * to get the SKL type widgets in that pipeline
+ */
+static int skl_tplg_alloc_pipe_widget(struct device *dev,
+ struct snd_soc_dapm_widget *w, struct skl_pipe *pipe)
+{
+ struct skl_module_cfg *src_module = NULL;
+ struct snd_soc_dapm_path *p = NULL;
+ struct skl_pipe_module *p_module = NULL;
+
+ p_module = devm_kzalloc(dev, sizeof(*p_module), GFP_KERNEL);
+ if (!p_module)
+ return -ENOMEM;
+
+ p_module->w = w;
+ list_add_tail(&p_module->node, &pipe->w_list);
+
+ snd_soc_dapm_widget_for_each_sink_path(w, p) {
+ if ((p->sink->priv == NULL)
+ && (!is_skl_dsp_widget_type(w)))
+ continue;
+
+ if ((p->sink->priv != NULL) && p->connect
+ && is_skl_dsp_widget_type(p->sink)) {
+
+ src_module = p->sink->priv;
+ if (pipe->ppl_id == src_module->pipe->ppl_id)
+ skl_tplg_alloc_pipe_widget(dev,
+ p->sink, pipe);
+ }
+ }
+ return 0;
+}
+
+/*
+ * Inside a pipe instance, we can have various modules. These modules need
+ * to instantiated in DSP by invoking INIT_MODULE IPC, which is achieved by
+ * skl_init_module() routine, so invoke that for all modules in a pipeline
+ */
+static int
+skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
+{
+ struct skl_pipe_module *w_module;
+ struct snd_soc_dapm_widget *w;
+ struct skl_module_cfg *mconfig;
+ struct skl_sst *ctx = skl->skl_sst;
+ int ret = 0;
+
+ list_for_each_entry(w_module, &pipe->w_list, node) {
+ w = w_module->w;
+ mconfig = w->priv;
+
+ /* check resource available */
+ if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+ return -ENOMEM;
+
+ /*
+ * apply fix/conversion to module params based on
+ * FE/BE params
+ */
+ skl_tplg_update_module_params(w, ctx);
+ ret = skl_init_module(ctx, mconfig, NULL);
+ if (ret < 0)
+ return ret;
+ }
+
+ return 0;
+}
+
+/*
+ * Mixer module represents a pipeline. So in the Pre-PMU event of mixer we
+ * need create the pipeline. So we do following:
+ * - check the resources
+ * - Create the pipeline
+ * - Initialize the modules in pipeline
+ * - finally bind all modules together
+ */
+static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
+ struct skl *skl)
+{
+ int ret;
+ struct skl_module_cfg *mconfig = w->priv;
+ struct skl_pipe_module *w_module;
+ struct skl_pipe *s_pipe = mconfig->pipe;
+ struct skl_module_cfg *src_module = NULL, *dst_module;
+ struct skl_sst *ctx = skl->skl_sst;
+
+ /* check resource available */
+ if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+ return -EBUSY;
+
+ if (!skl_tplg_alloc_pipe_mem(skl, mconfig))
+ return -ENOMEM;
+
+ /*
+ * Create a list of modules for pipe.
+ * This list contains modules from source to sink
+ */
+ ret = skl_create_pipeline(ctx, mconfig->pipe);
+ if (ret < 0)
+ return ret;
+
+ /*
+ * we create a w_list of all widgets in that pipe. This list is not
+ * freed on PMD event as widgets within a pipe are static. This
+ * saves us cycles to get widgets in pipe every time.
+ *
+ * So if we have already initialized all the widgets of a pipeline
+ * we skip, so check for list_empty and create the list if empty
+ */
+ if (list_empty(&s_pipe->w_list)) {
+ ret = skl_tplg_alloc_pipe_widget(ctx->dev, w, s_pipe);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* Init all pipe modules from source to sink */
+ ret = skl_tplg_init_pipe_modules(skl, s_pipe);
+ if (ret < 0)
+ return ret;
+
+ /* Bind modules from source to sink */
+ list_for_each_entry(w_module, &s_pipe->w_list, node) {
+ dst_module = w_module->w->priv;
+
+ if (src_module == NULL) {
+ src_module = dst_module;
+ continue;
+ }
+
+ ret = skl_bind_modules(ctx, src_module, dst_module);
+ if (ret < 0)
+ return ret;
+
+ src_module = dst_module;
+ }
+
+ return 0;
+}
+
+/*
+ * A PGA represents a module in a pipeline. So in the Pre-PMU event of PGA
+ * we need to do following:
+ * - Bind to sink pipeline
+ * Since the sink pipes can be running and we don't get mixer event on
+ * connect for already running mixer, we need to find the sink pipes
+ * here and bind to them. This way dynamic connect works.
+ * - Start sink pipeline, if not running
+ * - Then run current pipe
+ */
+static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
+ struct skl *skl)
+{
+ struct snd_soc_dapm_path *p;
+ struct skl_dapm_path_list *path_list;
+ struct snd_soc_dapm_widget *source, *sink;
+ struct skl_module_cfg *src_mconfig, *sink_mconfig;
+ struct skl_sst *ctx = skl->skl_sst;
+ int ret = 0;
+
+ source = w;
+ src_mconfig = source->priv;
+
+ /*
+ * find which sink it is connected to, bind with the sink,
+ * if sink is not started, start sink pipe first, then start
+ * this pipe
+ */
+ snd_soc_dapm_widget_for_each_source_path(w, p) {
+ if (!p->connect)
+ continue;
+
+ dev_dbg(ctx->dev, "%s: src widget=%s\n", __func__, w->name);
+ dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name);
+
+ /*
+ * here we will check widgets in sink pipelines, so that
+ * can be any widgets type and we are only interested if
+ * they are ones used for SKL so check that first
+ */
+ if ((p->sink->priv != NULL) &&
+ is_skl_dsp_widget_type(p->sink)) {
+
+ sink = p->sink;
+ src_mconfig = source->priv;
+ sink_mconfig = sink->priv;
+
+ /* Bind source to sink, mixin is always source */
+ ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig);
+ if (ret)
+ return ret;
+
+ /* Start sinks pipe first */
+ if (sink_mconfig->pipe->state != SKL_PIPE_STARTED) {
+ ret = skl_run_pipe(ctx, sink_mconfig->pipe);
+ if (ret)
+ return ret;
+ }
+
+ path_list = kzalloc(
+ sizeof(struct skl_dapm_path_list),
+ GFP_KERNEL);
+ if (path_list == NULL)
+ return -ENOMEM;
+
+ /* Add connected path to one global list */
+ path_list->dapm_path = p;
+ list_add_tail(&path_list->node, &skl->dapm_path_list);
+ break;
+ }
+ }
+
+ /* Start source pipe last after starting all sinks */
+ ret = skl_run_pipe(ctx, src_mconfig->pipe);
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * in the Post-PMU event of mixer we need to do following:
+ * - Check if this pipe is running
+ * - if not, then
+ * - bind this pipeline to its source pipeline
+ * if source pipe is already running, this means it is a dynamic
+ * connection and we need to bind only to that pipe
+ * - start this pipeline
+ */
+static int skl_tplg_mixer_dapm_post_pmu_event(struct snd_soc_dapm_widget *w,
+ struct skl *skl)
+{
+ int ret = 0;
+ struct snd_soc_dapm_path *p;
+ struct snd_soc_dapm_widget *source, *sink;
+ struct skl_module_cfg *src_mconfig, *sink_mconfig;
+ struct skl_sst *ctx = skl->skl_sst;
+ int src_pipe_started = 0;
+
+ sink = w;
+ sink_mconfig = sink->priv;
+
+ /*
+ * If source pipe is already started, that means source is driving
+ * one more sink before this sink got connected, Since source is
+ * started, bind this sink to source and start this pipe.
+ */
+ snd_soc_dapm_widget_for_each_sink_path(w, p) {
+ if (!p->connect)
+ continue;
+
+ dev_dbg(ctx->dev, "sink widget=%s\n", w->name);
+ dev_dbg(ctx->dev, "src widget=%s\n", p->source->name);
+
+ /*
+ * here we will check widgets in sink pipelines, so that
+ * can be any widgets type and we are only interested if
+ * they are ones used for SKL so check that first
+ */
+ if ((p->source->priv != NULL) &&
+ is_skl_dsp_widget_type(p->source)) {
+ source = p->source;
+ src_mconfig = source->priv;
+ sink_mconfig = sink->priv;
+ src_pipe_started = 1;
+
+ /*
+ * check pipe state, then no need to bind or start
+ * the pipe
+ */
+ if (src_mconfig->pipe->state != SKL_PIPE_STARTED)
+ src_pipe_started = 0;
+ }
+ }
+
+ if (src_pipe_started) {
+ ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig);
+ if (ret)
+ return ret;
+
+ ret = skl_run_pipe(ctx, sink_mconfig->pipe);
+ }
+
+ return ret;
+}
+
+/*
+ * in the Pre-PMD event of mixer we need to do following:
+ * - Stop the pipe
+ * - find the source connections and remove that from dapm_path_list
+ * - unbind with source pipelines if still connected
+ */
+static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
+ struct skl *skl)
+{
+ struct snd_soc_dapm_widget *source, *sink;
+ struct skl_module_cfg *src_mconfig, *sink_mconfig;
+ int ret = 0, path_found = 0;
+ struct skl_dapm_path_list *path_list, *tmp_list;
+ struct skl_sst *ctx = skl->skl_sst;
+
+ sink = w;
+ sink_mconfig = sink->priv;
+
+ /* Stop the pipe */
+ ret = skl_stop_pipe(ctx, sink_mconfig->pipe);
+ if (ret)
+ return ret;
+
+ /*
+ * This list, dapm_path_list handling here does not need any locks
+ * as we are under dapm lock while handling widget events.
+ * List can be manipulated safely only under dapm widgets handler
+ * routines
+ */
+ list_for_each_entry_safe(path_list, tmp_list,
+ &skl->dapm_path_list, node) {
+ if (path_list->dapm_path->sink == sink) {
+ dev_dbg(ctx->dev, "Path found = %s\n",
+ path_list->dapm_path->name);
+ source = path_list->dapm_path->source;
+ src_mconfig = source->priv;
+ path_found = 1;
+
+ list_del(&path_list->node);
+ kfree(path_list);
+ break;
+ }
+ }
+
+ /*
+ * If path_found == 1, that means pmd for source pipe has
+ * not occurred, source is connected to some other sink.
+ * so its responsibility of sink to unbind itself from source.
+ */
+ if (path_found) {
+ ret = skl_stop_pipe(ctx, src_mconfig->pipe);
+ if (ret < 0)
+ return ret;
+
+ ret = skl_unbind_modules(ctx, src_mconfig, sink_mconfig);
+ }
+
+ return ret;
+}
+
+/*
+ * in the Post-PMD event of mixer we need to do following:
+ * - Free the mcps used
+ * - Free the mem used
+ * - Unbind the modules within the pipeline
+ * - Delete the pipeline (modules are not required to be explicitly
+ * deleted, pipeline delete is enough here
+ */
+static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
+ struct skl *skl)
+{
+ struct skl_module_cfg *mconfig = w->priv;
+ struct skl_pipe_module *w_module;
+ struct skl_module_cfg *src_module = NULL, *dst_module;
+ struct skl_sst *ctx = skl->skl_sst;
+ struct skl_pipe *s_pipe = mconfig->pipe;
+ int ret = 0;
+
+ skl_tplg_free_pipe_mcps(skl, mconfig);
+
+ list_for_each_entry(w_module, &s_pipe->w_list, node) {
+ dst_module = w_module->w->priv;
+
+ if (src_module == NULL) {
+ src_module = dst_module;
+ continue;
+ }
+
+ ret = skl_unbind_modules(ctx, src_module, dst_module);
+ if (ret < 0)
+ return ret;
+
+ src_module = dst_module;
+ }
+
+ ret = skl_delete_pipe(ctx, mconfig->pipe);
+ skl_tplg_free_pipe_mem(skl, mconfig);
+
+ return ret;
+}
+
+/*
+ * in the Post-PMD event of PGA we need to do following:
+ * - Free the mcps used
+ * - Stop the pipeline
+ * - In source pipe is connected, unbind with source pipelines
+ */
+static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
+ struct skl *skl)
+{
+ struct snd_soc_dapm_widget *source, *sink;
+ struct skl_module_cfg *src_mconfig, *sink_mconfig;
+ int ret = 0, path_found = 0;
+ struct skl_dapm_path_list *path_list, *tmp_path_list;
+ struct skl_sst *ctx = skl->skl_sst;
+
+ source = w;
+ src_mconfig = source->priv;
+
+ skl_tplg_free_pipe_mcps(skl, src_mconfig);
+ /* Stop the pipe since this is a mixin module */
+ ret = skl_stop_pipe(ctx, src_mconfig->pipe);
+ if (ret)
+ return ret;
+
+ list_for_each_entry_safe(path_list, tmp_path_list, &skl->dapm_path_list, node) {
+ if (path_list->dapm_path->source == source) {
+ dev_dbg(ctx->dev, "Path found = %s\n",
+ path_list->dapm_path->name);
+ sink = path_list->dapm_path->sink;
+ sink_mconfig = sink->priv;
+ path_found = 1;
+
+ list_del(&path_list->node);
+ kfree(path_list);
+ break;
+ }
+ }
+
+ /*
+ * This is a connector and if path is found that means
+ * unbind between source and sink has not happened yet
+ */
+ if (path_found) {
+ ret = skl_stop_pipe(ctx, src_mconfig->pipe);
+ if (ret < 0)
+ return ret;
+
+ ret = skl_unbind_modules(ctx, src_mconfig, sink_mconfig);
+ }
+
+ return ret;
+}
+
+/*
+ * In modelling, we assume there will be ONLY one mixer in a pipeline. If
+ * mixer is not required then it is treated as static mixer aka vmixer with
+ * a hard path to source module
+ * So we don't need to check if source is started or not as hard path puts
+ * dependency on each other
+ */
+static int skl_tplg_vmixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct skl *skl = get_skl_ctx(dapm->dev);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return skl_tplg_mixer_dapm_pre_pmu_event(w, skl);
+
+ case SND_SOC_DAPM_POST_PMD:
+ return skl_tplg_mixer_dapm_post_pmd_event(w, skl);
+ }
+
+ return 0;
+}
+
+/*
+ * In modelling, we assume there will be ONLY one mixer in a pipeline. If a
+ * second one is required that is created as another pipe entity.
+ * The mixer is responsible for pipe management and represent a pipeline
+ * instance
+ */
+static int skl_tplg_mixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct skl *skl = get_skl_ctx(dapm->dev);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return skl_tplg_mixer_dapm_pre_pmu_event(w, skl);
+
+ case SND_SOC_DAPM_POST_PMU:
+ return skl_tplg_mixer_dapm_post_pmu_event(w, skl);
+
+ case SND_SOC_DAPM_PRE_PMD:
+ return skl_tplg_mixer_dapm_pre_pmd_event(w, skl);
+
+ case SND_SOC_DAPM_POST_PMD:
+ return skl_tplg_mixer_dapm_post_pmd_event(w, skl);
+ }
+
+ return 0;
+}
+
+/*
+ * In modelling, we assumed rest of the modules in pipeline are PGA. But we
+ * are interested in last PGA (leaf PGA) in a pipeline to disconnect with
+ * the sink when it is running (two FE to one BE or one FE to two BE)
+ * scenarios
+ */
+static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct skl *skl = get_skl_ctx(dapm->dev);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return skl_tplg_pga_dapm_pre_pmu_event(w, skl);
+
+ case SND_SOC_DAPM_POST_PMD:
+ return skl_tplg_pga_dapm_post_pmd_event(w, skl);
+ }
+
+ return 0;
+}
+
+/*
+ * The FE params are passed by hw_params of the DAI.
+ * On hw_params, the params are stored in Gateway module of the FE and we
+ * need to calculate the format in DSP module configuration, that
+ * conversion is done here
+ */
+int skl_tplg_update_pipe_params(struct device *dev,
+ struct skl_module_cfg *mconfig,
+ struct skl_pipe_params *params)
+{
+ struct skl_pipe *pipe = mconfig->pipe;
+ struct skl_module_fmt *format = NULL;
+
+ memcpy(pipe->p_params, params, sizeof(*params));
+
+ if (params->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ format = &mconfig->in_fmt;
+ else
+ format = &mconfig->out_fmt;
+
+ /* set the hw_params */
+ format->s_freq = params->s_freq;
+ format->channels = params->ch;
+ format->valid_bit_depth = skl_get_bit_depth(params->s_fmt);
+
+ /*
+ * 16 bit is 16 bit container whereas 24 bit is in 32 bit
+ * container so update bit depth accordingly
+ */
+ switch (format->valid_bit_depth) {
+ case SKL_DEPTH_16BIT:
+ format->bit_depth = format->valid_bit_depth;
+ break;
+
+ case SKL_DEPTH_24BIT:
+ format->bit_depth = SKL_DEPTH_32BIT;
+ break;
+
+ default:
+ dev_err(dev, "Invalid bit depth %x for pipe\n",
+ format->valid_bit_depth);
+ return -EINVAL;
+ }
+
+ if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mconfig->ibs = (format->s_freq / 1000) *
+ (format->channels) *
+ (format->bit_depth >> 3);
+ } else {
+ mconfig->obs = (format->s_freq / 1000) *
+ (format->channels) *
+ (format->bit_depth >> 3);
+ }
+
+ return 0;
+}
+
+/*
+ * Query the module config for the FE DAI
+ * This is used to find the hw_params set for that DAI and apply to FE
+ * pipeline
+ */
+struct skl_module_cfg *
+skl_tplg_fe_get_cpr_module(struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_dapm_widget *w;
+ struct snd_soc_dapm_path *p = NULL;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ w = dai->playback_widget;
+ snd_soc_dapm_widget_for_each_sink_path(w, p) {
+ if (p->connect && p->sink->power &&
+ is_skl_dsp_widget_type(p->sink))
+ continue;
+
+ if (p->sink->priv) {
+ dev_dbg(dai->dev, "set params for %s\n",
+ p->sink->name);
+ return p->sink->priv;
+ }
+ }
+ } else {
+ w = dai->capture_widget;
+ snd_soc_dapm_widget_for_each_source_path(w, p) {
+ if (p->connect && p->source->power &&
+ is_skl_dsp_widget_type(p->source))
+ continue;
+
+ if (p->source->priv) {
+ dev_dbg(dai->dev, "set params for %s\n",
+ p->source->name);
+ return p->source->priv;
+ }
+ }
+ }
+
+ return NULL;
+}
+
+static u8 skl_tplg_be_link_type(int dev_type)
+{
+ int ret;
+
+ switch (dev_type) {
+ case SKL_DEVICE_BT:
+ ret = NHLT_LINK_SSP;
+ break;
+
+ case SKL_DEVICE_DMIC:
+ ret = NHLT_LINK_DMIC;
+ break;
+
+ case SKL_DEVICE_I2S:
+ ret = NHLT_LINK_SSP;
+ break;
+
+ case SKL_DEVICE_HDALINK:
+ ret = NHLT_LINK_HDA;
+ break;
+
+ default:
+ ret = NHLT_LINK_INVALID;
+ break;
+ }
+
+ return ret;
+}
+
+/*
+ * Fill the BE gateway parameters
+ * The BE gateway expects a blob of parameters which are kept in the ACPI
+ * NHLT blob, so query the blob for interface type (i2s/pdm) and instance.
+ * The port can have multiple settings so pick based on the PCM
+ * parameters
+ */
+static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai,
+ struct skl_module_cfg *mconfig,
+ struct skl_pipe_params *params)
+{
+ struct skl_pipe *pipe = mconfig->pipe;
+ struct nhlt_specific_cfg *cfg;
+ struct skl *skl = get_skl_ctx(dai->dev);
+ int link_type = skl_tplg_be_link_type(mconfig->dev_type);
+
+ memcpy(pipe->p_params, params, sizeof(*params));
+
+ /* update the blob based on virtual bus_id*/
+ cfg = skl_get_ep_blob(skl, mconfig->vbus_id, link_type,
+ params->s_fmt, params->ch,
+ params->s_freq, params->stream);
+ if (cfg) {
+ mconfig->formats_config.caps_size = cfg->size;
+ mconfig->formats_config.caps = (u32 *) &cfg->caps;
+ } else {
+ dev_err(dai->dev, "Blob NULL for id %x type %d dirn %d\n",
+ mconfig->vbus_id, link_type,
+ params->stream);
+ dev_err(dai->dev, "PCM: ch %d, freq %d, fmt %d\n",
+ params->ch, params->s_freq, params->s_fmt);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int skl_tplg_be_set_src_pipe_params(struct snd_soc_dai *dai,
+ struct snd_soc_dapm_widget *w,
+ struct skl_pipe_params *params)
+{
+ struct snd_soc_dapm_path *p;
+ int ret = -EIO;
+
+ snd_soc_dapm_widget_for_each_source_path(w, p) {
+ if (p->connect && is_skl_dsp_widget_type(p->source) &&
+ p->source->priv) {
+
+ if (!p->source->power) {
+ ret = skl_tplg_be_fill_pipe_params(
+ dai, p->source->priv,
+ params);
+ if (ret < 0)
+ return ret;
+ } else {
+ return -EBUSY;
+ }
+ } else {
+ ret = skl_tplg_be_set_src_pipe_params(
+ dai, p->source, params);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static int skl_tplg_be_set_sink_pipe_params(struct snd_soc_dai *dai,
+ struct snd_soc_dapm_widget *w, struct skl_pipe_params *params)
+{
+ struct snd_soc_dapm_path *p = NULL;
+ int ret = -EIO;
+
+ snd_soc_dapm_widget_for_each_sink_path(w, p) {
+ if (p->connect && is_skl_dsp_widget_type(p->sink) &&
+ p->sink->priv) {
+
+ if (!p->sink->power) {
+ ret = skl_tplg_be_fill_pipe_params(
+ dai, p->sink->priv, params);
+ if (ret < 0)
+ return ret;
+ } else {
+ return -EBUSY;
+ }
+
+ } else {
+ ret = skl_tplg_be_set_sink_pipe_params(
+ dai, p->sink, params);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+/*
+ * BE hw_params can be a source parameters (capture) or sink parameters
+ * (playback). Based on sink and source we need to either find the source
+ * list or the sink list and set the pipeline parameters
+ */
+int skl_tplg_be_update_params(struct snd_soc_dai *dai,
+ struct skl_pipe_params *params)
+{
+ struct snd_soc_dapm_widget *w;
+
+ if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ w = dai->playback_widget;
+
+ return skl_tplg_be_set_src_pipe_params(dai, w, params);
+
+ } else {
+ w = dai->capture_widget;
+
+ return skl_tplg_be_set_sink_pipe_params(dai, w, params);
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_tplg_widget_events skl_tplg_widget_ops[] = {
+ {SKL_MIXER_EVENT, skl_tplg_mixer_event},
+ {SKL_VMIXER_EVENT, skl_tplg_vmixer_event},
+ {SKL_PGA_EVENT, skl_tplg_pga_event},
+};
+
+/*
+ * The topology binary passes the pin info for a module so initialize the pin
+ * info passed into module instance
+ */
+static void skl_fill_module_pin_info(struct skl_dfw_module_pin *dfw_pin,
+ struct skl_module_pin *m_pin,
+ bool is_dynamic, int max_pin)
+{
+ int i;
+
+ for (i = 0; i < max_pin; i++) {
+ m_pin[i].id.module_id = dfw_pin[i].module_id;
+ m_pin[i].id.instance_id = dfw_pin[i].instance_id;
+ m_pin[i].in_use = false;
+ m_pin[i].is_dynamic = is_dynamic;
+ }
+}
+
+/*
+ * Add pipeline from topology binary into driver pipeline list
+ *
+ * If already added we return that instance
+ * Otherwise we create a new instance and add into driver list
+ */
+static struct skl_pipe *skl_tplg_add_pipe(struct device *dev,
+ struct skl *skl, struct skl_dfw_pipe *dfw_pipe)
+{
+ struct skl_pipeline *ppl;
+ struct skl_pipe *pipe;
+ struct skl_pipe_params *params;
+
+ list_for_each_entry(ppl, &skl->ppl_list, node) {
+ if (ppl->pipe->ppl_id == dfw_pipe->pipe_id)
+ return ppl->pipe;
+ }
+
+ ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL);
+ if (!ppl)
+ return NULL;
+
+ pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL);
+ if (!pipe)
+ return NULL;
+
+ params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL);
+ if (!params)
+ return NULL;
+
+ pipe->ppl_id = dfw_pipe->pipe_id;
+ pipe->memory_pages = dfw_pipe->memory_pages;
+ pipe->pipe_priority = dfw_pipe->pipe_priority;
+ pipe->conn_type = dfw_pipe->conn_type;
+ pipe->state = SKL_PIPE_INVALID;
+ pipe->p_params = params;
+ INIT_LIST_HEAD(&pipe->w_list);
+
+ ppl->pipe = pipe;
+ list_add(&ppl->node, &skl->ppl_list);
+
+ return ppl->pipe;
+}
+
+/*
+ * Topology core widget load callback
+ *
+ * This is used to save the private data for each widget which gives
+ * information to the driver about module and pipeline parameters which DSP
+ * FW expects like ids, resource values, formats etc
+ */
+static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
+ struct snd_soc_dapm_widget *w,
+ struct snd_soc_tplg_dapm_widget *tplg_w)
+{
+ int ret;
+ struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt);
+ struct skl *skl = ebus_to_skl(ebus);
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct skl_module_cfg *mconfig;
+ struct skl_pipe *pipe;
+ struct skl_dfw_module *dfw_config =
+ (struct skl_dfw_module *)tplg_w->priv.data;
+
+ if (!tplg_w->priv.size)
+ goto bind_event;
+
+ mconfig = devm_kzalloc(bus->dev, sizeof(*mconfig), GFP_KERNEL);
+
+ if (!mconfig)
+ return -ENOMEM;
+
+ w->priv = mconfig;
+ mconfig->id.module_id = dfw_config->module_id;
+ mconfig->id.instance_id = dfw_config->instance_id;
+ mconfig->mcps = dfw_config->max_mcps;
+ mconfig->ibs = dfw_config->ibs;
+ mconfig->obs = dfw_config->obs;
+ mconfig->core_id = dfw_config->core_id;
+ mconfig->max_in_queue = dfw_config->max_in_queue;
+ mconfig->max_out_queue = dfw_config->max_out_queue;
+ mconfig->is_loadable = dfw_config->is_loadable;
+ mconfig->in_fmt.channels = dfw_config->in_fmt.channels;
+ mconfig->in_fmt.s_freq = dfw_config->in_fmt.freq;
+ mconfig->in_fmt.bit_depth = dfw_config->in_fmt.bit_depth;
+ mconfig->in_fmt.valid_bit_depth =
+ dfw_config->in_fmt.valid_bit_depth;
+ mconfig->in_fmt.ch_cfg = dfw_config->in_fmt.ch_cfg;
+ mconfig->out_fmt.channels = dfw_config->out_fmt.channels;
+ mconfig->out_fmt.s_freq = dfw_config->out_fmt.freq;
+ mconfig->out_fmt.bit_depth = dfw_config->out_fmt.bit_depth;
+ mconfig->out_fmt.valid_bit_depth =
+ dfw_config->out_fmt.valid_bit_depth;
+ mconfig->out_fmt.ch_cfg = dfw_config->out_fmt.ch_cfg;
+ mconfig->params_fixup = dfw_config->params_fixup;
+ mconfig->converter = dfw_config->converter;
+ mconfig->m_type = dfw_config->module_type;
+ mconfig->vbus_id = dfw_config->vbus_id;
+
+ pipe = skl_tplg_add_pipe(bus->dev, skl, &dfw_config->pipe);
+ if (pipe)
+ mconfig->pipe = pipe;
+
+ mconfig->dev_type = dfw_config->dev_type;
+ mconfig->hw_conn_type = dfw_config->hw_conn_type;
+ mconfig->time_slot = dfw_config->time_slot;
+ mconfig->formats_config.caps_size = dfw_config->caps.caps_size;
+
+ mconfig->m_in_pin = devm_kzalloc(bus->dev,
+ (mconfig->max_in_queue) *
+ sizeof(*mconfig->m_in_pin),
+ GFP_KERNEL);
+ if (!mconfig->m_in_pin)
+ return -ENOMEM;
+
+ mconfig->m_out_pin = devm_kzalloc(bus->dev, (mconfig->max_out_queue) *
+ sizeof(*mconfig->m_out_pin),
+ GFP_KERNEL);
+ if (!mconfig->m_out_pin)
+ return -ENOMEM;
+
+ skl_fill_module_pin_info(dfw_config->in_pin, mconfig->m_in_pin,
+ dfw_config->is_dynamic_in_pin,
+ mconfig->max_in_queue);
+
+ skl_fill_module_pin_info(dfw_config->out_pin, mconfig->m_out_pin,
+ dfw_config->is_dynamic_out_pin,
+ mconfig->max_out_queue);
+
+
+ if (mconfig->formats_config.caps_size == 0)
+ goto bind_event;
+
+ mconfig->formats_config.caps = (u32 *)devm_kzalloc(bus->dev,
+ mconfig->formats_config.caps_size, GFP_KERNEL);
+
+ if (mconfig->formats_config.caps == NULL)
+ return -ENOMEM;
+
+ memcpy(mconfig->formats_config.caps, dfw_config->caps.caps,
+ dfw_config->caps.caps_size);
+
+bind_event:
+ if (tplg_w->event_type == 0) {
+ dev_dbg(bus->dev, "ASoC: No event handler required\n");
+ return 0;
+ }
+
+ ret = snd_soc_tplg_widget_bind_event(w, skl_tplg_widget_ops,
+ ARRAY_SIZE(skl_tplg_widget_ops),
+ tplg_w->event_type);
+
+ if (ret) {
+ dev_err(bus->dev, "%s: No matching event handlers found for %d\n",
+ __func__, tplg_w->event_type);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_tplg_ops skl_tplg_ops = {
+ .widget_load = skl_tplg_widget_load,
+};
+
+/* This will be read from topology manifest, currently defined here */
+#define SKL_MAX_MCPS 30000000
+#define SKL_FW_MAX_MEM 1000000
+
+/*
+ * SKL topology init routine
+ */
+int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
+{
+ int ret;
+ const struct firmware *fw;
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct skl *skl = ebus_to_skl(ebus);
+
+ ret = request_firmware(&fw, "dfw_sst.bin", bus->dev);
+ if (ret < 0) {
+ dev_err(bus->dev, "tplg fw %s load failed with %d\n",
+ "dfw_sst.bin", ret);
+ return ret;
+ }
+
+ /*
+ * The complete tplg for SKL is loaded as index 0, we don't use
+ * any other index
+ */
+ ret = snd_soc_tplg_component_load(&platform->component,
+ &skl_tplg_ops, fw, 0);
+ if (ret < 0) {
+ dev_err(bus->dev, "tplg component load failed%d\n", ret);
+ return -EINVAL;
+ }
+
+ skl->resource.max_mcps = SKL_MAX_MCPS;
+ skl->resource.max_mem = SKL_FW_MAX_MEM;
+
+ return 0;
+}
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 8c7767b..76053a8 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -129,6 +129,11 @@
enum skl_s_freq src_cfg;
} __packed;
+struct notification_mask {
+ u32 notify;
+ u32 enable;
+} __packed;
+
struct skl_up_down_mixer_cfg {
struct skl_base_cfg base_cfg;
enum skl_ch_cfg out_ch_cfg;
@@ -153,8 +158,7 @@
union skl_ssp_dma_node {
u8 val;
struct {
- u8 dual_mono:1;
- u8 time_slot:3;
+ u8 time_slot_index:4;
u8 i2s_instance:4;
} dma_node;
};
@@ -263,6 +267,34 @@
struct skl_specific_cfg formats_config;
};
+struct skl_pipeline {
+ struct skl_pipe *pipe;
+ struct list_head node;
+};
+
+struct skl_dapm_path_list {
+ struct snd_soc_dapm_path *dapm_path;
+ struct list_head node;
+};
+
+static inline struct skl *get_skl_ctx(struct device *dev)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+
+ return ebus_to_skl(ebus);
+}
+
+int skl_tplg_be_update_params(struct snd_soc_dai *dai,
+ struct skl_pipe_params *params);
+void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai,
+ struct skl_pipe_params *params, int stream);
+int skl_tplg_init(struct snd_soc_platform *platform,
+ struct hdac_ext_bus *ebus);
+struct skl_module_cfg *skl_tplg_fe_get_cpr_module(
+ struct snd_soc_dai *dai, int stream);
+int skl_tplg_update_pipe_params(struct device *dev,
+ struct skl_module_cfg *mconfig, struct skl_pipe_params *params);
+
int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe);
int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h
index a506898..2bc396d 100644
--- a/sound/soc/intel/skylake/skl-tplg-interface.h
+++ b/sound/soc/intel/skylake/skl-tplg-interface.h
@@ -19,6 +19,29 @@
#ifndef __HDA_TPLG_INTERFACE_H__
#define __HDA_TPLG_INTERFACE_H__
+/*
+ * Default types range from 0~12. type can range from 0 to 0xff
+ * SST types start at higher to avoid any overlapping in future
+ */
+#define SOC_CONTROL_TYPE_HDA_SST_ALGO_PARAMS 0x100
+#define SOC_CONTROL_TYPE_HDA_SST_MUX 0x101
+#define SOC_CONTROL_TYPE_HDA_SST_MIX 0x101
+#define SOC_CONTROL_TYPE_HDA_SST_BYTE 0x103
+
+#define HDA_SST_CFG_MAX 900 /* size of copier cfg*/
+#define MAX_IN_QUEUE 8
+#define MAX_OUT_QUEUE 8
+
+/* Event types goes here */
+/* Reserve event type 0 for no event handlers */
+enum skl_event_types {
+ SKL_EVENT_NONE = 0,
+ SKL_MIXER_EVENT,
+ SKL_MUX_EVENT,
+ SKL_VMIXER_EVENT,
+ SKL_PGA_EVENT
+};
+
/**
* enum skl_ch_cfg - channel configuration
*
@@ -83,6 +106,66 @@
SKL_DEVICE_I2S = 0x2,
SKL_DEVICE_SLIMBUS = 0x3,
SKL_DEVICE_HDALINK = 0x4,
+ SKL_DEVICE_HDAHOST = 0x5,
SKL_DEVICE_NONE
};
+
+struct skl_dfw_module_pin {
+ u16 module_id;
+ u16 instance_id;
+} __packed;
+
+struct skl_dfw_module_fmt {
+ u32 channels;
+ u32 freq;
+ u32 bit_depth;
+ u32 valid_bit_depth;
+ u32 ch_cfg;
+} __packed;
+
+struct skl_dfw_module_caps {
+ u32 caps_size;
+ u32 caps[HDA_SST_CFG_MAX];
+};
+
+struct skl_dfw_pipe {
+ u8 pipe_id;
+ u8 pipe_priority;
+ u16 conn_type;
+ u32 memory_pages;
+} __packed;
+
+struct skl_dfw_module {
+ u16 module_id;
+ u16 instance_id;
+ u32 max_mcps;
+ u8 core_id;
+ u8 max_in_queue;
+ u8 max_out_queue;
+ u8 is_loadable;
+ u8 conn_type;
+ u8 dev_type;
+ u8 hw_conn_type;
+ u8 time_slot;
+ u32 obs;
+ u32 ibs;
+ u32 params_fixup;
+ u32 converter;
+ u32 module_type;
+ u32 vbus_id;
+ u8 is_dynamic_in_pin;
+ u8 is_dynamic_out_pin;
+ struct skl_dfw_pipe pipe;
+ struct skl_dfw_module_fmt in_fmt;
+ struct skl_dfw_module_fmt out_fmt;
+ struct skl_dfw_module_pin in_pin[MAX_IN_QUEUE];
+ struct skl_dfw_module_pin out_pin[MAX_OUT_QUEUE];
+ struct skl_dfw_module_caps caps;
+} __packed;
+
+struct skl_dfw_algo_data {
+ u32 max;
+ char *params;
+} __packed;
+
#endif
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 348d094..5319529 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -166,12 +166,20 @@
struct pci_dev *pci = to_pci_dev(dev);
struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct skl *skl = ebus_to_skl(ebus);
+ int ret;
dev_dbg(bus->dev, "in %s\n", __func__);
/* enable controller wake up event */
snd_hdac_chip_updatew(bus, WAKEEN, 0, STATESTS_INT_MASK);
+ snd_hdac_ext_bus_link_power_down_all(ebus);
+
+ ret = skl_suspend_dsp(skl);
+ if (ret < 0)
+ return ret;
+
snd_hdac_bus_stop_chip(bus);
snd_hdac_bus_enter_link_reset(bus);
@@ -183,7 +191,7 @@
struct pci_dev *pci = to_pci_dev(dev);
struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
struct hdac_bus *bus = ebus_to_hbus(ebus);
- struct skl *hda = ebus_to_skl(ebus);
+ struct skl *skl = ebus_to_skl(ebus);
int status;
dev_dbg(bus->dev, "in %s\n", __func__);
@@ -191,12 +199,12 @@
/* Read STATESTS before controller reset */
status = snd_hdac_chip_readw(bus, STATESTS);
- skl_init_pci(hda);
+ skl_init_pci(skl);
snd_hdac_bus_init_chip(bus, true);
/* disable controller Wake Up event */
snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, 0);
- return 0;
+ return skl_resume_dsp(skl);
}
#endif /* CONFIG_PM */
@@ -453,21 +461,28 @@
if (err < 0)
goto out_free;
+ skl->nhlt = skl_nhlt_init(bus->dev);
+
+ if (skl->nhlt == NULL)
+ goto out_free;
+
pci_set_drvdata(skl->pci, ebus);
/* check if dsp is there */
if (ebus->ppcap) {
- /* TODO register with dsp IPC */
- dev_dbg(bus->dev, "Register dsp\n");
+ err = skl_init_dsp(skl);
+ if (err < 0) {
+ dev_dbg(bus->dev, "error failed to register dsp\n");
+ goto out_free;
+ }
}
-
if (ebus->mlcap)
snd_hdac_ext_bus_get_ml_capabilities(ebus);
/* create device for soc dmic */
err = skl_dmic_device_register(skl);
if (err < 0)
- goto out_free;
+ goto out_dsp_free;
/* register platform dai and controls */
err = skl_platform_register(bus->dev);
@@ -491,6 +506,8 @@
skl_platform_unregister(bus->dev);
out_dmic_free:
skl_dmic_device_unregister(skl);
+out_dsp_free:
+ skl_free_dsp(skl);
out_free:
skl->init_failed = 1;
skl_free(ebus);
@@ -507,6 +524,7 @@
pm_runtime_get_noresume(&pci->dev);
pci_dev_put(pci);
skl_platform_unregister(&pci->dev);
+ skl_free_dsp(skl);
skl_dmic_device_unregister(skl);
skl_free(ebus);
dev_set_drvdata(&pci->dev, NULL);
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index f7fdbb0..dd2e79a 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -48,6 +48,13 @@
#define AZX_REG_VS_SDXEFIFOS_XBASE 0x1094
#define AZX_REG_VS_SDXEFIFOS_XINTERVAL 0x20
+struct skl_dsp_resource {
+ u32 max_mcps;
+ u32 max_mem;
+ u32 mcps;
+ u32 mem;
+};
+
struct skl {
struct hdac_ext_bus ebus;
struct pci_dev *pci;
@@ -55,8 +62,12 @@
unsigned int init_failed:1; /* delayed init failed */
struct platform_device *dmic_dev;
- void __iomem *nhlt; /* nhlt ptr */
+ void *nhlt; /* nhlt ptr */
struct skl_sst *skl_sst; /* sst skl ctx */
+
+ struct skl_dsp_resource resource;
+ struct list_head ppl_list;
+ struct list_head dapm_path_list;
};
#define skl_to_ebus(s) (&(s)->ebus)
@@ -72,8 +83,8 @@
int skl_platform_unregister(struct device *dev);
int skl_platform_register(struct device *dev);
-void __iomem *skl_nhlt_init(struct device *dev);
-void skl_nhlt_free(void __iomem *addr);
+void *skl_nhlt_init(struct device *dev);
+void skl_nhlt_free(void *addr);
struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance,
u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn);
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index b05fb1c..794a349 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -485,6 +485,7 @@
{ .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 },
{ /* sentinel */ }
};
+MODULE_DEVICE_TABLE(of, jz4740_of_matches);
#endif
static int jz4740_i2s_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
index de7563b..e0304d5 100644
--- a/sound/soc/kirkwood/armada-370-db.c
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -130,6 +130,7 @@
{ .compatible = "marvell,a370db-audio" },
{ },
};
+MODULE_DEVICE_TABLE(of, a370db_dt_ids);
static struct platform_driver a370db_driver = {
.driver = {
diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c
index 684e8a7..71a1a35 100644
--- a/sound/soc/mediatek/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173-max98090.c
@@ -179,21 +179,13 @@
}
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
-static int mt8173_max98090_dev_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static const struct of_device_id mt8173_max98090_dt_match[] = {
{ .compatible = "mediatek,mt8173-max98090", },
{ }
@@ -209,7 +201,6 @@
#endif
},
.probe = mt8173_max98090_dev_probe,
- .remove = mt8173_max98090_dev_remove,
};
module_platform_driver(mt8173_max98090_driver);
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 86cf975..50ba538 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -246,21 +246,13 @@
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
-static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = {
{ .compatible = "mediatek,mt8173-rt5650-rt5676", },
{ }
@@ -276,7 +268,6 @@
#endif
},
.probe = mt8173_rt5650_rt5676_dev_probe,
- .remove = mt8173_rt5650_rt5676_dev_remove,
};
module_platform_driver(mt8173_rt5650_rt5676_driver);
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 6e6fce6..2b23ffb 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -142,7 +142,7 @@
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
ret);
@@ -154,12 +154,8 @@
static int mxs_sgtl5000_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
mxs_saif_put_mclk(0);
- snd_soc_unregister_card(card);
-
return 0;
}
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index dcb5336..190f868 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -99,8 +99,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
n810_ext_control(&rtd->card->dapm);
return clk_prepare_enable(sys_clkout2);
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 3bebfb1..5e21f08 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -107,8 +107,7 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
rx51_ext_control(&card->dapm);
return 0;
@@ -297,7 +296,7 @@
dev_err(card->dev, "Failed to add TPA6130A2 controls\n");
return err;
}
- snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
+ snd_soc_limit_volume(card, "TPA6130A2 Headphone Playback Volume", 42);
err = omap_mcbsp_st_add_controls(rtd, 2);
if (err < 0) {
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 2b26318..6147e86 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -116,26 +116,19 @@
int ret;
brownstone.dev = &pdev->dev;
- ret = snd_soc_register_card(&brownstone);
+ ret = devm_snd_soc_register_card(&pdev->dev, &brownstone);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int brownstone_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&brownstone);
- return 0;
-}
-
static struct platform_driver mmp_driver = {
.driver = {
.name = "brownstone-audio",
.pm = &snd_soc_pm_ops,
},
.probe = brownstone_probe,
- .remove = brownstone_remove,
};
module_platform_driver(mmp_driver);
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 3580d10..c97dc13 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -295,28 +295,19 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int corgi_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver corgi_driver = {
.driver = {
.name = "corgi-audio",
.pm = &snd_soc_pm_ops,
},
.probe = corgi_probe,
- .remove = corgi_remove,
};
module_platform_driver(corgi_driver);
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index d72e124..1de8765 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -138,7 +138,7 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -149,10 +149,7 @@
static int e740_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 48f2d7c..b7eb7cd 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -120,7 +120,7 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -131,10 +131,7 @@
static int e750_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 45d4bd4..41bf714 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -119,7 +119,7 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -130,10 +130,7 @@
static int e800_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 9f8be7c..ecbf287 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -193,7 +193,7 @@
return ret;
snd_soc_card_hx4700.dev = &pdev->dev;
- ret = snd_soc_register_card(&snd_soc_card_hx4700);
+ ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700);
if (ret)
gpio_free_array(hx4700_audio_gpios,
ARRAY_SIZE(hx4700_audio_gpios));
@@ -203,8 +203,6 @@
static int hx4700_audio_remove(struct platform_device *pdev)
{
- snd_soc_unregister_card(&snd_soc_card_hx4700);
-
gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index 29fabbf..9d0e407 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -72,28 +72,19 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int imote2_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver imote2_driver = {
.driver = {
.name = "imote2-audio",
.pm = &snd_soc_pm_ops,
},
.probe = imote2_probe,
- .remove = imote2_remove,
};
module_platform_driver(imote2_driver);
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index a9615a57..29bc60e8 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -181,7 +181,7 @@
return -ENODEV;
mioa701.dev = &pdev->dev;
- rc = snd_soc_register_card(&mioa701);
+ rc = devm_snd_soc_register_card(&pdev->dev, &mioa701);
if (!rc)
dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
"lead to overheating and possible destruction of your device."
@@ -189,17 +189,8 @@
return rc;
}
-static int mioa701_wm9713_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver mioa701_wm9713_driver = {
.probe = mioa701_wm9713_probe,
- .remove = mioa701_wm9713_remove,
.driver = {
.name = "mioa701-wm9713",
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index c20bbc0..4e74d95 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -140,22 +140,15 @@
palm27x_asoc.dev = &pdev->dev;
- ret = snd_soc_register_card(&palm27x_asoc);
+ ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int palm27x_asoc_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_card(&palm27x_asoc);
- return 0;
-}
-
static struct platform_driver palm27x_wm9712_driver = {
.probe = palm27x_asoc_probe,
- .remove = palm27x_asoc_remove,
.driver = {
.name = "palm27x-asoc",
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 80b457a..84d0e2e 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -267,28 +267,19 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
-static int poodle_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver poodle_driver = {
.driver = {
.name = "poodle-audio",
.pm = &snd_soc_pm_ops,
},
.probe = poodle_probe,
- .remove = poodle_remove,
};
module_platform_driver(poodle_driver);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 3da485e..da03fad 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -809,6 +809,7 @@
{ .compatible = "mrvl,pxa-ssp-dai" },
{}
};
+MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids);
#endif
static int asoc_ssp_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 9e4b04e..f3de615 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -15,6 +15,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
#include <sound/core.h>
#include <sound/ac97_codec.h>
@@ -49,7 +50,11 @@
.reset = pxa2xx_ac97_cold_reset,
};
-static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
+static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 11,
+};
+
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -57,7 +62,11 @@
.filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
};
-static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
+static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 12,
+};
+
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -65,7 +74,10 @@
.filter_data = &pxa2xx_ac97_pcm_stereo_out_req,
};
-static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10;
+static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 10,
+};
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
.addr = __PREG(MODR),
.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
@@ -73,7 +85,10 @@
.filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req,
};
-static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9;
+static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 9,
+};
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
.addr = __PREG(MODR),
.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
@@ -81,7 +96,10 @@
.filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req,
};
-static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8;
+static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 8,
+};
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
.addr = __PREG(MCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
@@ -89,9 +107,8 @@
.filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req,
};
-static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *cpu_dai)
+static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
{
struct snd_dmaengine_dai_dma_data *dma_data;
@@ -105,9 +122,8 @@
return 0;
}
-static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *cpu_dai)
+static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
{
struct snd_dmaengine_dai_dma_data *dma_data;
@@ -121,9 +137,8 @@
return 0;
}
-static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *cpu_dai)
+static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
{
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
@@ -139,15 +154,15 @@
SNDRV_PCM_RATE_48000)
static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
- .hw_params = pxa2xx_ac97_hw_params,
+ .startup = pxa2xx_ac97_hifi_startup,
};
static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
- .hw_params = pxa2xx_ac97_hw_aux_params,
+ .startup = pxa2xx_ac97_aux_startup,
};
static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
- .hw_params = pxa2xx_ac97_hw_mic_params,
+ .startup = pxa2xx_ac97_mic_startup,
};
/*
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 6b4e400..0389cf7 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -319,6 +319,9 @@
/* Along with FIFO servicing */
SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
+ snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out,
+ &pxa2xx_i2s_pcm_stereo_in);
+
return 0;
}
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 831ee37..9f39039 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -15,8 +15,6 @@
#include <linux/dmaengine.h>
#include <linux/of.h>
-#include <mach/dma.h>
-
#include <sound/core.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
@@ -27,11 +25,8 @@
static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_dmaengine_dai_dma_data *dma;
- int ret;
dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
@@ -40,40 +35,13 @@
if (!dma)
return 0;
- /* this may get called several times by oss emulation
- * with different params */
- if (prtd->params == NULL) {
- prtd->params = dma;
- ret = pxa_request_dma("name", DMA_PRIO_LOW,
- pxa2xx_pcm_dma_irq, substream);
- if (ret < 0)
- return ret;
- prtd->dma_ch = ret;
- } else if (prtd->params != dma) {
- pxa_free_dma(prtd->dma_ch);
- prtd->params = dma;
- ret = pxa_request_dma("name", DMA_PRIO_LOW,
- pxa2xx_pcm_dma_irq, substream);
- if (ret < 0)
- return ret;
- prtd->dma_ch = ret;
- }
-
return __pxa2xx_pcm_hw_params(substream, params);
}
static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
- struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
-
__pxa2xx_pcm_hw_free(substream);
- if (prtd->dma_ch >= 0) {
- pxa_free_dma(prtd->dma_ch);
- prtd->dma_ch = -1;
- prtd->params = NULL;
- }
-
return 0;
}
@@ -132,6 +100,7 @@
{ .compatible = "mrvl,pxa-pcm-audio" },
{ }
};
+MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match);
#endif
static struct platform_driver pxa_pcm_driver = {
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 461123a..b002226 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -305,7 +305,7 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -322,9 +322,6 @@
static int spitz_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
gpio_free(spitz_mic_gpio);
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f59f566..49518dd 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -233,7 +233,7 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -244,10 +244,7 @@
static int tosa_remove(struct platform_device *pdev)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
gpio_free(TOSA_GPIO_L_MUTE);
- snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index 1753c7d..65c20f7 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -128,7 +128,7 @@
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
@@ -136,22 +136,12 @@
return ret;
}
-static int ttc_dkb_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
-}
-
static struct platform_driver ttc_dkb_driver = {
.driver = {
.name = "ttc-dkb-audio",
.pm = &snd_soc_pm_ops,
},
.probe = ttc_dkb_probe,
- .remove = ttc_dkb_remove,
};
module_platform_driver(ttc_dkb_driver);
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 97bc202..e5101e0 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -438,7 +438,8 @@
if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) {
dev_err(&pdev->dev,
"%s() error getting mi2s-bit-clk: %ld\n",
- __func__, PTR_ERR(drvdata->mi2s_bit_clk[i]));
+ __func__,
+ PTR_ERR(drvdata->mi2s_bit_clk[dai_id]));
return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]);
}
}
diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
index 58bae8e..f1e0c70 100644
--- a/sound/soc/rockchip/Kconfig
+++ b/sound/soc/rockchip/Kconfig
@@ -15,9 +15,17 @@
Rockchip I2S device. The device supports upto maximum of
8 channels each for play and record.
+config SND_SOC_ROCKCHIP_SPDIF
+ tristate "Rockchip SPDIF Device Driver"
+ depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ help
+ Say Y or M if you want to add support for SPDIF driver for
+ Rockchip SPDIF transceiver device.
+
config SND_SOC_ROCKCHIP_MAX98090
tristate "ASoC support for Rockchip boards using a MAX98090 codec"
- depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+ depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
select SND_SOC_ROCKCHIP_I2S
select SND_SOC_MAX98090
select SND_SOC_TS3A227E
@@ -27,7 +35,7 @@
config SND_SOC_ROCKCHIP_RT5645
tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec"
- depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+ depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
select SND_SOC_ROCKCHIP_I2S
select SND_SOC_RT5645
help
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
index 1bc1dc3..c0bf560 100644
--- a/sound/soc/rockchip/Makefile
+++ b/sound/soc/rockchip/Makefile
@@ -1,7 +1,9 @@
# ROCKCHIP Platform Support
-snd-soc-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-spdif-objs := rockchip_spdif.o
-obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o
snd-soc-rockchip-max98090-objs := rockchip_max98090.o
snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index b936102..58ee645 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -226,6 +226,7 @@
struct snd_soc_dai *dai)
{
struct rk_i2s_dev *i2s = to_info(dai);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
unsigned int val = 0;
switch (params_format(params)) {
@@ -245,13 +246,46 @@
return -EINVAL;
}
- regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val);
- regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val);
+ switch (params_channels(params)) {
+ case 8:
+ val |= I2S_CHN_8;
+ break;
+ case 6:
+ val |= I2S_CHN_6;
+ break;
+ case 4:
+ val |= I2S_CHN_4;
+ break;
+ case 2:
+ val |= I2S_CHN_2;
+ break;
+ default:
+ dev_err(i2s->dev, "invalid channel: %d\n",
+ params_channels(params));
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ regmap_update_bits(i2s->regmap, I2S_RXCR,
+ I2S_RXCR_VDW_MASK | I2S_RXCR_CSR_MASK,
+ val);
+ else
+ regmap_update_bits(i2s->regmap, I2S_TXCR,
+ I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK,
+ val);
+
regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK,
I2S_DMACR_TDL(16));
regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK,
I2S_DMACR_RDL(16));
+ val = I2S_CKR_TRCM_TXRX;
+ if (dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates)
+ val = I2S_CKR_TRCM_TXSHARE;
+
+ regmap_update_bits(i2s->regmap, I2S_CKR,
+ I2S_CKR_TRCM_MASK,
+ val);
return 0;
}
@@ -415,10 +449,12 @@
static int rockchip_i2s_probe(struct platform_device *pdev)
{
+ struct device_node *node = pdev->dev.of_node;
struct rk_i2s_dev *i2s;
struct resource *res;
void __iomem *regs;
int ret;
+ int val;
i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
if (!i2s) {
@@ -475,6 +511,14 @@
goto err_pm_disable;
}
+ /* refine capture channels */
+ if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) {
+ if (val >= 2 && val <= 8)
+ rockchip_i2s_dai.capture.channels_max = val;
+ else
+ rockchip_i2s_dai.capture.channels_max = 2;
+ }
+
ret = devm_snd_soc_register_component(&pdev->dev,
&rockchip_i2s_component,
&rockchip_i2s_dai, 1);
diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h
index 93f456f..dc6e2c7 100644
--- a/sound/soc/rockchip/rockchip_i2s.h
+++ b/sound/soc/rockchip/rockchip_i2s.h
@@ -49,6 +49,9 @@
* RXCR
* receive operation control register
*/
+#define I2S_RXCR_CSR_SHIFT 15
+#define I2S_RXCR_CSR(x) (x << I2S_RXCR_CSR_SHIFT)
+#define I2S_RXCR_CSR_MASK (3 << I2S_RXCR_CSR_SHIFT)
#define I2S_RXCR_HWT BIT(14)
#define I2S_RXCR_SJM_SHIFT 12
#define I2S_RXCR_SJM_R (0 << I2S_RXCR_SJM_SHIFT)
@@ -75,6 +78,12 @@
* CKR
* clock generation register
*/
+#define I2S_CKR_TRCM_SHIFT 28
+#define I2S_CKR_TRCM(x) (x << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_TXRX (0 << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_TXSHARE (1 << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_RXSHARE (2 << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_MASK (3 << I2S_CKR_TRCM_SHIFT)
#define I2S_CKR_MSS_SHIFT 27
#define I2S_CKR_MSS_MASTER (0 << I2S_CKR_MSS_SHIFT)
#define I2S_CKR_MSS_SLAVE (1 << I2S_CKR_MSS_SHIFT)
@@ -207,6 +216,13 @@
ROCKCHIP_DIV_BCLK,
};
+/* channel select */
+#define I2S_CSR_SHIFT 15
+#define I2S_CHN_2 (0 << I2S_CSR_SHIFT)
+#define I2S_CHN_4 (1 << I2S_CSR_SHIFT)
+#define I2S_CHN_6 (2 << I2S_CSR_SHIFT)
+#define I2S_CHN_8 (3 << I2S_CSR_SHIFT)
+
/* I2S REGS */
#define I2S_TXCR (0x0000)
#define I2S_RXCR (0x0004)
diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c
new file mode 100644
index 0000000..a38a302
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_spdif.c
@@ -0,0 +1,405 @@
+/* sound/soc/rockchip/rk_spdif.c
+ *
+ * ALSA SoC Audio Layer - Rockchip I2S Controller driver
+ *
+ * Copyright (c) 2014 Rockchip Electronics Co. Ltd.
+ * Author: Jianqun <jay.xu@rock-chips.com>
+ * Copyright (c) 2015 Collabora Ltd.
+ * Author: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/of_gpio.h>
+#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <linux/mfd/syscon.h>
+#include <linux/regmap.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "rockchip_spdif.h"
+
+enum rk_spdif_type {
+ RK_SPDIF_RK3066,
+ RK_SPDIF_RK3188,
+ RK_SPDIF_RK3288,
+};
+
+#define RK3288_GRF_SOC_CON2 0x24c
+
+struct rk_spdif_dev {
+ struct device *dev;
+
+ struct clk *mclk;
+ struct clk *hclk;
+
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+
+ struct regmap *regmap;
+};
+
+static const struct of_device_id rk_spdif_match[] = {
+ { .compatible = "rockchip,rk3066-spdif",
+ .data = (void *) RK_SPDIF_RK3066 },
+ { .compatible = "rockchip,rk3188-spdif",
+ .data = (void *) RK_SPDIF_RK3188 },
+ { .compatible = "rockchip,rk3288-spdif",
+ .data = (void *) RK_SPDIF_RK3288 },
+ {},
+};
+MODULE_DEVICE_TABLE(of, rk_spdif_match);
+
+static int rk_spdif_runtime_suspend(struct device *dev)
+{
+ struct rk_spdif_dev *spdif = dev_get_drvdata(dev);
+
+ clk_disable_unprepare(spdif->mclk);
+ clk_disable_unprepare(spdif->hclk);
+
+ return 0;
+}
+
+static int rk_spdif_runtime_resume(struct device *dev)
+{
+ struct rk_spdif_dev *spdif = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(spdif->mclk);
+ if (ret) {
+ dev_err(spdif->dev, "mclk clock enable failed %d\n", ret);
+ return ret;
+ }
+
+ ret = clk_prepare_enable(spdif->hclk);
+ if (ret) {
+ dev_err(spdif->dev, "hclk clock enable failed %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int rk_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai);
+ unsigned int val = SPDIF_CFGR_HALFWORD_ENABLE;
+ int srate, mclk;
+ int ret;
+
+ srate = params_rate(params);
+ switch (srate) {
+ case 32000:
+ case 48000:
+ case 96000:
+ mclk = 96000 * 128; /* 12288000 hz */
+ break;
+ case 44100:
+ mclk = 44100 * 256; /* 11289600 hz */
+ break;
+ case 192000:
+ mclk = 192000 * 128; /* 24576000 hz */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val |= SPDIF_CFGR_VDW_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val |= SPDIF_CFGR_VDW_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val |= SPDIF_CFGR_VDW_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Set clock and calculate divider */
+ ret = clk_set_rate(spdif->mclk, mclk);
+ if (ret != 0) {
+ dev_err(spdif->dev, "Failed to set module clock rate: %d\n",
+ ret);
+ return ret;
+ }
+
+ val |= SPDIF_CFGR_CLK_DIV(mclk/(srate * 256));
+ ret = regmap_update_bits(spdif->regmap, SPDIF_CFGR,
+ SPDIF_CFGR_CLK_DIV_MASK | SPDIF_CFGR_HALFWORD_ENABLE |
+ SDPIF_CFGR_VDW_MASK,
+ val);
+
+ return ret;
+}
+
+static int rk_spdif_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai);
+ int ret;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR,
+ SPDIF_DMACR_TDE_ENABLE,
+ SPDIF_DMACR_TDE_ENABLE);
+
+ if (ret != 0)
+ return ret;
+
+ ret = regmap_update_bits(spdif->regmap, SPDIF_XFER,
+ SPDIF_XFER_TXS_START,
+ SPDIF_XFER_TXS_START);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR,
+ SPDIF_DMACR_TDE_ENABLE,
+ SPDIF_DMACR_TDE_DISABLE);
+
+ if (ret != 0)
+ return ret;
+
+ ret = regmap_update_bits(spdif->regmap, SPDIF_XFER,
+ SPDIF_XFER_TXS_START,
+ SPDIF_XFER_TXS_STOP);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int rk_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+ struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai);
+
+ dai->playback_dma_data = &spdif->playback_dma_data;
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops rk_spdif_dai_ops = {
+ .hw_params = rk_spdif_hw_params,
+ .trigger = rk_spdif_trigger,
+};
+
+static struct snd_soc_dai_driver rk_spdif_dai = {
+ .probe = rk_spdif_dai_probe,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ },
+ .ops = &rk_spdif_dai_ops,
+};
+
+static const struct snd_soc_component_driver rk_spdif_component = {
+ .name = "rockchip-spdif",
+};
+
+static bool rk_spdif_wr_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case SPDIF_CFGR:
+ case SPDIF_DMACR:
+ case SPDIF_INTCR:
+ case SPDIF_XFER:
+ case SPDIF_SMPDR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rk_spdif_rd_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case SPDIF_CFGR:
+ case SPDIF_SDBLR:
+ case SPDIF_INTCR:
+ case SPDIF_INTSR:
+ case SPDIF_XFER:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rk_spdif_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case SPDIF_INTSR:
+ case SPDIF_SDBLR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config rk_spdif_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = SPDIF_SMPDR,
+ .writeable_reg = rk_spdif_wr_reg,
+ .readable_reg = rk_spdif_rd_reg,
+ .volatile_reg = rk_spdif_volatile_reg,
+ .cache_type = REGCACHE_FLAT,
+};
+
+static int rk_spdif_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct rk_spdif_dev *spdif;
+ const struct of_device_id *match;
+ struct resource *res;
+ void __iomem *regs;
+ int ret;
+
+ match = of_match_node(rk_spdif_match, np);
+ if ((int) match->data == RK_SPDIF_RK3288) {
+ struct regmap *grf;
+
+ grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf");
+ if (IS_ERR(grf)) {
+ dev_err(&pdev->dev,
+ "rockchip_spdif missing 'rockchip,grf' \n");
+ return PTR_ERR(grf);
+ }
+
+ /* Select the 8 channel SPDIF solution on RK3288 as
+ * the 2 channel one does not appear to work
+ */
+ regmap_write(grf, RK3288_GRF_SOC_CON2, BIT(1) << 16);
+ }
+
+ spdif = devm_kzalloc(&pdev->dev, sizeof(*spdif), GFP_KERNEL);
+ if (!spdif)
+ return -ENOMEM;
+
+ spdif->hclk = devm_clk_get(&pdev->dev, "hclk");
+ if (IS_ERR(spdif->hclk)) {
+ dev_err(&pdev->dev, "Can't retrieve rk_spdif bus clock\n");
+ return PTR_ERR(spdif->hclk);
+ }
+ ret = clk_prepare_enable(spdif->hclk);
+ if (ret) {
+ dev_err(spdif->dev, "hclock enable failed %d\n", ret);
+ return ret;
+ }
+
+ spdif->mclk = devm_clk_get(&pdev->dev, "mclk");
+ if (IS_ERR(spdif->mclk)) {
+ dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n");
+ return PTR_ERR(spdif->mclk);
+ }
+
+ ret = clk_prepare_enable(spdif->mclk);
+ if (ret) {
+ dev_err(spdif->dev, "clock enable failed %d\n", ret);
+ return ret;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs,
+ &rk_spdif_regmap_config);
+ if (IS_ERR(spdif->regmap)) {
+ dev_err(&pdev->dev,
+ "Failed to initialise managed register map\n");
+ return PTR_ERR(spdif->regmap);
+ }
+
+ spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR;
+ spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ spdif->playback_dma_data.maxburst = 4;
+
+ spdif->dev = &pdev->dev;
+ dev_set_drvdata(&pdev->dev, spdif);
+
+ pm_runtime_set_active(&pdev->dev);
+ pm_runtime_enable(&pdev->dev);
+ pm_request_idle(&pdev->dev);
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &rk_spdif_component,
+ &rk_spdif_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Could not register DAI\n");
+ goto err_pm_runtime;
+ }
+
+ ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (ret) {
+ dev_err(&pdev->dev, "Could not register PCM\n");
+ goto err_pm_runtime;
+ }
+
+ return 0;
+
+err_pm_runtime:
+ pm_runtime_disable(&pdev->dev);
+
+ return ret;
+}
+
+static int rk_spdif_remove(struct platform_device *pdev)
+{
+ struct rk_spdif_dev *spdif = dev_get_drvdata(&pdev->dev);
+
+ pm_runtime_disable(&pdev->dev);
+ if (!pm_runtime_status_suspended(&pdev->dev))
+ rk_spdif_runtime_suspend(&pdev->dev);
+
+ clk_disable_unprepare(spdif->mclk);
+ clk_disable_unprepare(spdif->hclk);
+
+ return 0;
+}
+
+static const struct dev_pm_ops rk_spdif_pm_ops = {
+ SET_RUNTIME_PM_OPS(rk_spdif_runtime_suspend, rk_spdif_runtime_resume,
+ NULL)
+};
+
+static struct platform_driver rk_spdif_driver = {
+ .probe = rk_spdif_probe,
+ .remove = rk_spdif_remove,
+ .driver = {
+ .name = "rockchip-spdif",
+ .of_match_table = of_match_ptr(rk_spdif_match),
+ .pm = &rk_spdif_pm_ops,
+ },
+};
+module_platform_driver(rk_spdif_driver);
+
+MODULE_ALIAS("platform:rockchip-spdif");
+MODULE_DESCRIPTION("ROCKCHIP SPDIF transceiver Interface");
+MODULE_AUTHOR("Sjoerd Simons <sjoerd.simons@collabora.co.uk>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h
new file mode 100644
index 0000000..07f86a2
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_spdif.h
@@ -0,0 +1,63 @@
+/*
+ * ALSA SoC Audio Layer - Rockchip SPDIF transceiver driver
+ *
+ * Copyright (c) 2015 Collabora Ltd.
+ * Author: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
+ *
+ * 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 _ROCKCHIP_SPDIF_H
+#define _ROCKCHIP_SPDIF_H
+
+/*
+ * CFGR
+ * transfer configuration register
+*/
+#define SPDIF_CFGR_CLK_DIV_SHIFT (16)
+#define SPDIF_CFGR_CLK_DIV_MASK (0xff << SPDIF_CFGR_CLK_DIV_SHIFT)
+#define SPDIF_CFGR_CLK_DIV(x) (x << SPDIF_CFGR_CLK_DIV_SHIFT)
+
+#define SPDIF_CFGR_HALFWORD_SHIFT 2
+#define SPDIF_CFGR_HALFWORD_DISABLE (0 << SPDIF_CFGR_HALFWORD_SHIFT)
+#define SPDIF_CFGR_HALFWORD_ENABLE (1 << SPDIF_CFGR_HALFWORD_SHIFT)
+
+#define SPDIF_CFGR_VDW_SHIFT 0
+#define SPDIF_CFGR_VDW(x) (x << SPDIF_CFGR_VDW_SHIFT)
+#define SDPIF_CFGR_VDW_MASK (0xf << SPDIF_CFGR_VDW_SHIFT)
+
+#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x00)
+#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x01)
+#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x10)
+
+/*
+ * DMACR
+ * DMA control register
+*/
+#define SPDIF_DMACR_TDE_SHIFT 5
+#define SPDIF_DMACR_TDE_DISABLE (0 << SPDIF_DMACR_TDE_SHIFT)
+#define SPDIF_DMACR_TDE_ENABLE (1 << SPDIF_DMACR_TDE_SHIFT)
+
+#define SPDIF_DMACR_TDL_SHIFT 0
+#define SPDIF_DMACR_TDL(x) ((x) << SPDIF_DMACR_TDL_SHIFT)
+#define SPDIF_DMACR_TDL_MASK (0x1f << SDPIF_DMACR_TDL_SHIFT)
+
+/*
+ * XFER
+ * Transfer control register
+*/
+#define SPDIF_XFER_TXS_SHIFT 0
+#define SPDIF_XFER_TXS_STOP (0 << SPDIF_XFER_TXS_SHIFT)
+#define SPDIF_XFER_TXS_START (1 << SPDIF_XFER_TXS_SHIFT)
+
+#define SPDIF_CFGR (0x0000)
+#define SPDIF_SDBLR (0x0004)
+#define SPDIF_DMACR (0x0008)
+#define SPDIF_INTCR (0x000c)
+#define SPDIF_INTSR (0x0010)
+#define SPDIF_XFER (0x0018)
+#define SPDIF_SMPDR (0x0020)
+
+#endif /* _ROCKCHIP_SPDIF_H */
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index c72e9fb..5f5825f 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -26,16 +26,15 @@
#include <mach/gpio-samsung.h>
#include "s3c24xx-i2s.h"
-static unsigned int rates[] = {
+static const unsigned int rates[] = {
11025,
22050,
44100,
};
-static struct snd_pcm_hw_constraint_list hw_rates = {
+static const struct snd_pcm_hw_constraint_list hw_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
- .mask = 0,
};
static struct snd_soc_jack hp_jack;
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 35e37c4..fa096ab 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -38,16 +38,15 @@
static int rx1950_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
-static unsigned int rates[] = {
+static const unsigned int rates[] = {
16000,
44100,
48000,
};
-static struct snd_pcm_hw_constraint_list hw_rates = {
+static const struct snd_pcm_hw_constraint_list hw_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
- .mask = 0,
};
static struct snd_soc_jack hp_jack;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 07114b0..206d1ed 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,10 +37,11 @@
config SND_SOC_RCAR
tristate "R-Car series SRU/SCU/SSIU/SSI support"
depends on DMA_OF
+ depends on COMMON_CLK
select SND_SIMPLE_CARD
select REGMAP_MMIO
help
- This option enables R-Car SUR/SCU/SSIU/SSI sound support
+ This option enables R-Car SRU/SCU/SSIU/SSI sound support
config SND_SOC_RSRC_CARD
tristate "Renesas Sampling Rate Convert Sound Card"
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index fefc881..2a5b3a2 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -7,7 +7,7 @@
* License. See the file "COPYING" in the main directory of this archive
* for more details.
*/
-#include <linux/sh_clk.h>
+#include <linux/clk-provider.h>
#include "rsnd.h"
#define CLKA 0
@@ -16,12 +16,26 @@
#define CLKI 3
#define CLKMAX 4
+#define CLKOUT 0
+#define CLKOUT1 1
+#define CLKOUT2 2
+#define CLKOUT3 3
+#define CLKOUTMAX 4
+
+#define BRRx_MASK(x) (0x3FF & x)
+
+static struct rsnd_mod_ops adg_ops = {
+ .name = "adg",
+};
+
struct rsnd_adg {
struct clk *clk[CLKMAX];
+ struct clk *clkout[CLKOUTMAX];
+ struct clk_onecell_data onecell;
+ struct rsnd_mod mod;
- int rbga_rate_for_441khz_div_6; /* RBGA */
- int rbgb_rate_for_48khz_div_6; /* RBGB */
- u32 ckr;
+ int rbga_rate_for_441khz; /* RBGA */
+ int rbgb_rate_for_48khz; /* RBGB */
};
#define for_each_rsnd_clk(pos, adg, i) \
@@ -29,17 +43,36 @@
(i < CLKMAX) && \
((pos) = adg->clk[i]); \
i++)
+#define for_each_rsnd_clkout(pos, adg, i) \
+ for (i = 0; \
+ (i < CLKOUTMAX) && \
+ ((pos) = adg->clkout[i]); \
+ i++)
#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
+static u32 rsnd_adg_calculate_rbgx(unsigned long div)
+{
+ int i, ratio;
+
+ if (!div)
+ return 0;
+
+ for (i = 3; i >= 0; i--) {
+ ratio = 2 << (i * 2);
+ if (0 == (div % ratio))
+ return (u32)((i << 8) | ((div / ratio) - 1));
+ }
+
+ return ~0;
+}
static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
{
struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io);
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
int id = rsnd_mod_id(mod);
int ws = id;
- if (rsnd_ssi_is_pin_sharing(rsnd_ssi_mod_get(priv, id))) {
+ if (rsnd_ssi_is_pin_sharing(io)) {
switch (id) {
case 1:
case 2:
@@ -60,6 +93,9 @@
int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io)
{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
int id = rsnd_mod_id(mod);
int shift = (id % 2) ? 16 : 0;
u32 mask, val;
@@ -69,21 +105,26 @@
val = val << shift;
mask = 0xffff << shift;
- rsnd_mod_bset(mod, CMDOUT_TIMSEL, mask, val);
+ rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val);
return 0;
}
-static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod,
+static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io,
u32 timsel)
{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
int is_play = rsnd_io_is_play(io);
- int id = rsnd_mod_id(mod);
+ int id = rsnd_mod_id(src_mod);
int shift = (id % 2) ? 16 : 0;
u32 mask, ws;
u32 in, out;
+ rsnd_mod_confirm_src(src_mod);
+
ws = rsnd_adg_ssi_ws_timing_gen2(io);
in = (is_play) ? timsel : ws;
@@ -95,37 +136,38 @@
switch (id / 2) {
case 0:
- rsnd_mod_bset(mod, SRCIN_TIMSEL0, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL0, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out);
break;
case 1:
- rsnd_mod_bset(mod, SRCIN_TIMSEL1, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL1, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out);
break;
case 2:
- rsnd_mod_bset(mod, SRCIN_TIMSEL2, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL2, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out);
break;
case 3:
- rsnd_mod_bset(mod, SRCIN_TIMSEL3, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL3, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out);
break;
case 4:
- rsnd_mod_bset(mod, SRCIN_TIMSEL4, mask, in);
- rsnd_mod_bset(mod, SRCOUT_TIMSEL4, mask, out);
+ rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4, mask, in);
+ rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out);
break;
}
return 0;
}
-int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io,
unsigned int src_rate,
unsigned int dst_rate)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
struct device *dev = rsnd_priv_to_dev(priv);
int idx, sel, div, step, ret;
u32 val, en;
@@ -134,10 +176,12 @@
clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */
clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */
clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */
- adg->rbga_rate_for_441khz_div_6,/* 0011: RBGA */
- adg->rbgb_rate_for_48khz_div_6, /* 0100: RBGB */
+ adg->rbga_rate_for_441khz, /* 0011: RBGA */
+ adg->rbgb_rate_for_48khz, /* 0100: RBGB */
};
+ rsnd_mod_confirm_src(src_mod);
+
min = ~0;
val = 0;
en = 0;
@@ -175,25 +219,27 @@
return -EIO;
}
- ret = rsnd_adg_set_src_timsel_gen2(mod, io, val);
+ ret = rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
if (ret < 0) {
dev_err(dev, "timsel error\n");
return ret;
}
- rsnd_mod_bset(mod, DIV_EN, en, en);
+ rsnd_mod_bset(adg_mod, DIV_EN, en, en);
dev_dbg(dev, "convert rate %d <-> %d\n", src_rate, dst_rate);
return 0;
}
-int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *src_mod,
struct rsnd_dai_stream *io)
{
u32 val = rsnd_adg_ssi_ws_timing_gen2(io);
- return rsnd_adg_set_src_timsel_gen2(mod, io, val);
+ rsnd_mod_confirm_src(src_mod);
+
+ return rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
}
int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
@@ -202,6 +248,7 @@
unsigned int dst_rate)
{
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
struct device *dev = rsnd_priv_to_dev(priv);
int idx, sel, div, shift;
u32 mask, val;
@@ -211,8 +258,8 @@
clk_get_rate(adg->clk[CLKB]), /* 001: CLKB */
clk_get_rate(adg->clk[CLKC]), /* 010: CLKC */
0, /* 011: MLBCLK (not used) */
- adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */
- adg->rbgb_rate_for_48khz_div_6, /* 101: RBGB */
+ adg->rbga_rate_for_441khz, /* 100: RBGA */
+ adg->rbgb_rate_for_48khz, /* 101: RBGB */
};
/* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */
@@ -238,13 +285,13 @@
switch (id / 4) {
case 0:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL3, mask, val);
break;
case 1:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL4, mask, val);
break;
case 2:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL5, mask, val);
break;
}
@@ -257,12 +304,17 @@
return 0;
}
-static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
+static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val)
{
- int id = rsnd_mod_id(mod);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+ int id = rsnd_mod_id(ssi_mod);
int shift = (id % 4) * 8;
u32 mask = 0xFF << shift;
+ rsnd_mod_confirm_ssi(ssi_mod);
+
val = val << shift;
/*
@@ -274,13 +326,13 @@
switch (id / 4) {
case 0:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val);
break;
case 1:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val);
break;
case 2:
- rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val);
+ rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val);
break;
}
}
@@ -326,14 +378,14 @@
}
/*
- * find 1/6 clock from BRGA/BRGB
+ * find divided clock from BRGA/BRGB
*/
- if (rate == adg->rbga_rate_for_441khz_div_6) {
+ if (rate == adg->rbga_rate_for_441khz) {
data = 0x10;
goto found_clock;
}
- if (rate == adg->rbgb_rate_for_48khz_div_6) {
+ if (rate == adg->rbgb_rate_for_48khz) {
data = 0x20;
goto found_clock;
}
@@ -342,29 +394,60 @@
found_clock:
- /* see rsnd_adg_ssi_clk_init() */
- rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr);
- rsnd_mod_write(mod, BRRA, 0x00000002); /* 1/6 */
- rsnd_mod_write(mod, BRRB, 0x00000002); /* 1/6 */
-
/*
* This "mod" = "ssi" here.
* we can get "ssi id" from mod
*/
rsnd_adg_set_ssi_clk(mod, data);
- dev_dbg(dev, "ADG: ssi%d selects clk%d = %d",
- rsnd_mod_id(mod), i, rate);
+ dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod),
+ data, rate);
return 0;
}
-static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
+static void rsnd_adg_get_clkin(struct rsnd_priv *priv,
+ struct rsnd_adg *adg)
+{
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct clk *clk;
+ static const char * const clk_name[] = {
+ [CLKA] = "clk_a",
+ [CLKB] = "clk_b",
+ [CLKC] = "clk_c",
+ [CLKI] = "clk_i",
+ };
+ int i;
+
+ for (i = 0; i < CLKMAX; i++) {
+ clk = devm_clk_get(dev, clk_name[i]);
+ adg->clk[i] = IS_ERR(clk) ? NULL : clk;
+ }
+
+ for_each_rsnd_clk(clk, adg, i)
+ dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+}
+
+static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
+ struct rsnd_adg *adg)
{
struct clk *clk;
- unsigned long rate;
- u32 ckr;
+ struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct device_node *np = dev->of_node;
+ u32 ckr, rbgx, rbga, rbgb;
+ u32 rate, req_rate, div;
+ uint32_t count = 0;
+ unsigned long req_48kHz_rate, req_441kHz_rate;
int i;
+ const char *parent_clk_name = NULL;
+ static const char * const clkout_name[] = {
+ [CLKOUT] = "audio_clkout",
+ [CLKOUT1] = "audio_clkout1",
+ [CLKOUT2] = "audio_clkout2",
+ [CLKOUT3] = "audio_clkout3",
+ };
int brg_table[] = {
[CLKA] = 0x0,
[CLKB] = 0x1,
@@ -372,19 +455,34 @@
[CLKI] = 0x2,
};
+ of_property_read_u32(np, "#clock-cells", &count);
+
+ /*
+ * ADG supports BRRA/BRRB output only
+ * this means all clkout0/1/2/3 will be same rate
+ */
+ of_property_read_u32(np, "clock-frequency", &req_rate);
+ req_48kHz_rate = 0;
+ req_441kHz_rate = 0;
+ if (0 == (req_rate % 44100))
+ req_441kHz_rate = req_rate;
+ if (0 == (req_rate % 48000))
+ req_48kHz_rate = req_rate;
+
/*
* This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC
* have 44.1kHz or 48kHz base clocks for now.
*
* SSI itself can divide parent clock by 1/1 - 1/16
- * So, BRGA outputs 44.1kHz base parent clock 1/32,
- * and, BRGB outputs 48.0kHz base parent clock 1/32 here.
* see
* rsnd_adg_ssi_clk_try_start()
+ * rsnd_ssi_master_clk_start()
*/
ckr = 0;
- adg->rbga_rate_for_441khz_div_6 = 0;
- adg->rbgb_rate_for_48khz_div_6 = 0;
+ rbga = 2; /* default 1/6 */
+ rbgb = 2; /* default 1/6 */
+ adg->rbga_rate_for_441khz = 0;
+ adg->rbgb_rate_for_48khz = 0;
for_each_rsnd_clk(clk, adg, i) {
rate = clk_get_rate(clk);
@@ -392,19 +490,86 @@
continue;
/* RBGA */
- if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) {
- adg->rbga_rate_for_441khz_div_6 = rate / 6;
- ckr |= brg_table[i] << 20;
+ if (!adg->rbga_rate_for_441khz && (0 == rate % 44100)) {
+ div = 6;
+ if (req_441kHz_rate)
+ div = rate / req_441kHz_rate;
+ rbgx = rsnd_adg_calculate_rbgx(div);
+ if (BRRx_MASK(rbgx) == rbgx) {
+ rbga = rbgx;
+ adg->rbga_rate_for_441khz = rate / div;
+ ckr |= brg_table[i] << 20;
+ if (req_441kHz_rate)
+ parent_clk_name = __clk_get_name(clk);
+ }
}
/* RBGB */
- if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) {
- adg->rbgb_rate_for_48khz_div_6 = rate / 6;
- ckr |= brg_table[i] << 16;
+ if (!adg->rbgb_rate_for_48khz && (0 == rate % 48000)) {
+ div = 6;
+ if (req_48kHz_rate)
+ div = rate / req_48kHz_rate;
+ rbgx = rsnd_adg_calculate_rbgx(div);
+ if (BRRx_MASK(rbgx) == rbgx) {
+ rbgb = rbgx;
+ adg->rbgb_rate_for_48khz = rate / div;
+ ckr |= brg_table[i] << 16;
+ if (req_48kHz_rate) {
+ parent_clk_name = __clk_get_name(clk);
+ ckr |= 0x80000000;
+ }
+ }
}
}
- adg->ckr = ckr;
+ /*
+ * ADG supports BRRA/BRRB output only.
+ * this means all clkout0/1/2/3 will be * same rate
+ */
+
+ /*
+ * for clkout
+ */
+ if (!count) {
+ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT],
+ parent_clk_name,
+ (parent_clk_name) ?
+ 0 : CLK_IS_ROOT, req_rate);
+ if (!IS_ERR(clk)) {
+ adg->clkout[CLKOUT] = clk;
+ of_clk_add_provider(np, of_clk_src_simple_get, clk);
+ }
+ }
+ /*
+ * for clkout0/1/2/3
+ */
+ else {
+ for (i = 0; i < CLKOUTMAX; i++) {
+ clk = clk_register_fixed_rate(dev, clkout_name[i],
+ parent_clk_name,
+ (parent_clk_name) ?
+ 0 : CLK_IS_ROOT,
+ req_rate);
+ if (!IS_ERR(clk)) {
+ adg->onecell.clks = adg->clkout;
+ adg->onecell.clk_num = CLKOUTMAX;
+
+ adg->clkout[i] = clk;
+
+ of_clk_add_provider(np, of_clk_src_onecell_get,
+ &adg->onecell);
+ }
+ }
+ }
+
+ rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr);
+ rsnd_mod_write(adg_mod, BRRA, rbga);
+ rsnd_mod_write(adg_mod, BRRB, rbgb);
+
+ for_each_rsnd_clkout(clk, adg, i)
+ dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+ dev_dbg(dev, "SSICKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
+ ckr, rbga, rbgb);
}
int rsnd_adg_probe(struct platform_device *pdev,
@@ -413,8 +578,6 @@
{
struct rsnd_adg *adg;
struct device *dev = rsnd_priv_to_dev(priv);
- struct clk *clk;
- int i;
adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
if (!adg) {
@@ -422,15 +585,16 @@
return -ENOMEM;
}
- adg->clk[CLKA] = devm_clk_get(dev, "clk_a");
- adg->clk[CLKB] = devm_clk_get(dev, "clk_b");
- adg->clk[CLKC] = devm_clk_get(dev, "clk_c");
- adg->clk[CLKI] = devm_clk_get(dev, "clk_i");
+ /*
+ * ADG is special module.
+ * Use ADG mod without rsnd_mod_init() to make debug easy
+ * for rsnd_write/rsnd_read
+ */
+ adg->mod.ops = &adg_ops;
+ adg->mod.priv = priv;
- for_each_rsnd_clk(clk, adg, i)
- dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
-
- rsnd_adg_ssi_clk_init(priv, adg);
+ rsnd_adg_get_clkin(priv, adg);
+ rsnd_adg_get_clkout(priv, adg);
priv->adg = adg;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f3feed5..deed48e 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -110,6 +110,7 @@
static const struct of_device_id rsnd_of_match[] = {
{ .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 },
{ .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 },
+ { .compatible = "renesas,rcar_sound-gen3", .data = &rsnd_of_data_gen2 }, /* gen2 compatible */
{},
};
MODULE_DEVICE_TABLE(of, rsnd_of_match);
@@ -126,6 +127,17 @@
#define rsnd_info_id(priv, io, name) \
((io)->info->name - priv->info->name##_info)
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type)
+{
+ if (mod->type != type) {
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct device *dev = rsnd_priv_to_dev(priv);
+
+ dev_warn(dev, "%s[%d] is not your expected module\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
+ }
+}
+
/*
* rsnd_mod functions
*/
@@ -288,7 +300,7 @@
/*
* rsnd_dai functions
*/
-#define __rsnd_mod_call(mod, io, func, param...) \
+#define rsnd_mod_call(mod, io, func, param...) \
({ \
struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \
struct device *dev = rsnd_priv_to_dev(priv); \
@@ -296,24 +308,17 @@
u8 val = (mod->status >> __rsnd_mod_shift_##func) & 0xF; \
u8 add = ((val + __rsnd_mod_add_##func) & 0xF); \
int ret = 0; \
- int called = 0; \
- if (val == __rsnd_mod_call_##func) { \
- called = 1; \
- ret = (mod)->ops->func(mod, io, param); \
- } \
+ int call = (val == __rsnd_mod_call_##func) && (mod)->ops->func; \
mod->status = (mod->status & ~mask) + \
(add << __rsnd_mod_shift_##func); \
- dev_dbg(dev, "%s[%d] 0x%08x %s\n", \
- rsnd_mod_name(mod), rsnd_mod_id(mod), mod->status, \
- called ? #func : ""); \
+ dev_dbg(dev, "%s[%d]\t0x%08x %s\n", \
+ rsnd_mod_name(mod), rsnd_mod_id(mod), \
+ mod->status, call ? #func : ""); \
+ if (call) \
+ ret = (mod)->ops->func(mod, io, param); \
ret; \
})
-#define rsnd_mod_call(mod, io, func, param...) \
- (!(mod) ? -ENODEV : \
- !((mod)->ops->func) ? 0 : \
- __rsnd_mod_call(mod, io, func, param))
-
#define rsnd_dai_call(fn, io, param...) \
({ \
struct rsnd_mod *mod; \
@@ -322,9 +327,7 @@
mod = (io)->mod[i]; \
if (!mod) \
continue; \
- ret = rsnd_mod_call(mod, io, fn, param); \
- if (ret < 0) \
- break; \
+ ret |= rsnd_mod_call(mod, io, fn, param); \
} \
ret; \
})
@@ -490,16 +493,10 @@
break;
case SNDRV_PCM_TRIGGER_STOP:
ret = rsnd_dai_call(stop, io, priv);
- if (ret < 0)
- goto dai_trigger_end;
- ret = rsnd_dai_call(quit, io, priv);
- if (ret < 0)
- goto dai_trigger_end;
+ ret |= rsnd_dai_call(quit, io, priv);
- ret = rsnd_platform_call(priv, dai, stop, ssi_id);
- if (ret < 0)
- goto dai_trigger_end;
+ ret |= rsnd_platform_call(priv, dai, stop, ssi_id);
rsnd_dai_stream_quit(io);
break;
@@ -1224,20 +1221,11 @@
};
int ret, i;
- info = NULL;
- of_data = NULL;
- if (of_id) {
- info = devm_kzalloc(&pdev->dev,
- sizeof(struct rcar_snd_info), GFP_KERNEL);
- of_data = of_id->data;
- } else {
- info = pdev->dev.platform_data;
- }
-
- if (!info) {
- dev_err(dev, "driver needs R-Car sound information\n");
- return -ENODEV;
- }
+ info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info),
+ GFP_KERNEL);
+ if (!info)
+ return -ENOMEM;
+ of_data = of_id->data;
/*
* init priv data
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 05498bb..3cb214a 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -35,7 +35,7 @@
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- rsnd_mod_hw_start(mod);
+ rsnd_mod_power_on(mod);
rsnd_ctu_initialize_lock(mod);
@@ -50,7 +50,7 @@
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- rsnd_mod_hw_stop(mod);
+ rsnd_mod_power_off(mod);
return 0;
}
@@ -66,7 +66,7 @@
if (WARN_ON(id < 0 || id >= rsnd_ctu_nr(priv)))
id = 0;
- return &((struct rsnd_ctu *)(priv->ctu) + id)->mod;
+ return rsnd_mod_get((struct rsnd_ctu *)(priv->ctu) + id);
}
static void rsnd_of_parse_ctu(struct platform_device *pdev,
@@ -118,10 +118,8 @@
int i, nr, ret;
/* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv)) {
- dev_warn(dev, "CTU is not supported on Gen1\n");
- return -EINVAL;
- }
+ if (rsnd_is_gen1(priv))
+ return 0;
rsnd_of_parse_ctu(pdev, of_data, priv);
@@ -150,7 +148,7 @@
ctu->info = &info->ctu_info[i];
- ret = rsnd_mod_init(priv, &ctu->mod, &rsnd_ctu_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops,
clk, RSND_MOD_CTU, i);
if (ret)
return ret;
@@ -166,6 +164,6 @@
int i;
for_each_rsnd_ctu(ctu, priv, i) {
- rsnd_mod_quit(&ctu->mod);
+ rsnd_mod_quit(rsnd_mod_get(ctu));
}
}
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index bfbb8a5..5d084d0 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -470,7 +470,7 @@
dev_err(dev, "DVC is selected without SRC\n");
/* use SSIU or SSI ? */
- if (is_ssi && rsnd_ssi_use_busif(io, mod))
+ if (is_ssi && rsnd_ssi_use_busif(io))
is_ssi++;
return (is_from) ?
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index 5779638..58f6909 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -153,7 +153,7 @@
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- rsnd_mod_hw_start(mod);
+ rsnd_mod_power_on(mod);
rsnd_dvc_soft_reset(mod);
@@ -175,7 +175,7 @@
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- rsnd_mod_hw_stop(mod);
+ rsnd_mod_power_off(mod);
return 0;
}
@@ -282,7 +282,7 @@
if (WARN_ON(id < 0 || id >= rsnd_dvc_nr(priv)))
id = 0;
- return &((struct rsnd_dvc *)(priv->dvc) + id)->mod;
+ return rsnd_mod_get((struct rsnd_dvc *)(priv->dvc) + id);
}
static void rsnd_of_parse_dvc(struct platform_device *pdev,
@@ -333,10 +333,8 @@
int i, nr, ret;
/* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv)) {
- dev_warn(dev, "CMD is not supported on Gen1\n");
- return -EINVAL;
- }
+ if (rsnd_is_gen1(priv))
+ return 0;
rsnd_of_parse_dvc(pdev, of_data, priv);
@@ -361,7 +359,7 @@
dvc->info = &info->dvc_info[i];
- ret = rsnd_mod_init(priv, &dvc->mod, &rsnd_dvc_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops,
clk, RSND_MOD_DVC, i);
if (ret)
return ret;
@@ -377,6 +375,6 @@
int i;
for_each_rsnd_dvc(dvc, priv, i) {
- rsnd_mod_quit(&dvc->mod);
+ rsnd_mod_quit(rsnd_mod_get(dvc));
}
}
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index f04d17b..76da762 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -22,13 +22,15 @@
#include "rsnd.h"
struct rsnd_gen {
- void __iomem *base[RSND_BASE_MAX];
-
struct rsnd_gen_ops *ops;
+ /* RSND_BASE_MAX base */
+ void __iomem *base[RSND_BASE_MAX];
+ phys_addr_t res[RSND_BASE_MAX];
struct regmap *regmap[RSND_BASE_MAX];
+
+ /* RSND_REG_MAX base */
struct regmap_field *regs[RSND_REG_MAX];
- phys_addr_t res[RSND_REG_MAX];
};
#define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen)
@@ -79,11 +81,11 @@
if (!rsnd_is_accessible_reg(priv, gen, reg))
return 0;
+ regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val);
+
dev_dbg(dev, "r %s[%d] - %4d : %08x\n",
rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val);
- regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val);
-
return val;
}
@@ -182,6 +184,7 @@
if (IS_ERR(regmap))
return PTR_ERR(regmap);
+ /* RSND_BASE_MAX base */
gen->base[reg_id] = base;
gen->regmap[reg_id] = regmap;
gen->res[reg_id] = res->start;
@@ -198,6 +201,7 @@
if (IS_ERR(regs))
return PTR_ERR(regs);
+ /* RSND_REG_MAX base */
gen->regs[conf[i].idx] = regs;
}
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 0d5c102..953dd0b 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -58,7 +58,7 @@
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- rsnd_mod_hw_start(mod);
+ rsnd_mod_power_on(mod);
rsnd_mix_soft_reset(mod);
@@ -83,7 +83,7 @@
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- rsnd_mod_hw_stop(mod);
+ rsnd_mod_power_off(mod);
return 0;
}
@@ -99,7 +99,7 @@
if (WARN_ON(id < 0 || id >= rsnd_mix_nr(priv)))
id = 0;
- return &((struct rsnd_mix *)(priv->mix) + id)->mod;
+ return rsnd_mod_get((struct rsnd_mix *)(priv->mix) + id);
}
static void rsnd_of_parse_mix(struct platform_device *pdev,
@@ -151,10 +151,8 @@
int i, nr, ret;
/* This driver doesn't support Gen1 at this point */
- if (rsnd_is_gen1(priv)) {
- dev_warn(dev, "MIX is not supported on Gen1\n");
- return -EINVAL;
- }
+ if (rsnd_is_gen1(priv))
+ return 0;
rsnd_of_parse_mix(pdev, of_data, priv);
@@ -179,7 +177,7 @@
mix->info = &info->mix_info[i];
- ret = rsnd_mod_init(priv, &mix->mod, &rsnd_mix_ops,
+ ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops,
clk, RSND_MOD_MIX, i);
if (ret)
return ret;
@@ -195,6 +193,6 @@
int i;
for_each_rsnd_mix(mix, priv, i) {
- rsnd_mod_quit(&mix->mod);
+ rsnd_mod_quit(rsnd_mod_get(mix));
}
}
diff --git a/include/sound/rcar_snd.h b/sound/soc/sh/rcar/rcar_snd.h
similarity index 98%
rename from include/sound/rcar_snd.h
rename to sound/soc/sh/rcar/rcar_snd.h
index bb7b2eb..d8e33d3 100644
--- a/include/sound/rcar_snd.h
+++ b/sound/soc/sh/rcar/rcar_snd.h
@@ -12,7 +12,6 @@
#ifndef RCAR_SND_H
#define RCAR_SND_H
-#include <linux/sh_clk.h>
#define RSND_GEN1_SRU 0
#define RSND_GEN1_ADG 1
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 7a0e52b..0853298 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -21,10 +21,11 @@
#include <linux/of_irq.h>
#include <linux/sh_dma.h>
#include <linux/workqueue.h>
-#include <sound/rcar_snd.h>
#include <sound/soc.h>
#include <sound/pcm_params.h>
+#include "rcar_snd.h"
+
/*
* pseudo register
*
@@ -214,6 +215,7 @@
};
#define rsnd_dma_to_dmaen(dma) (&(dma)->dma.en)
#define rsnd_dma_to_dmapp(dma) (&(dma)->dma.pp)
+#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
void rsnd_dma_start(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
void rsnd_dma_stop(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
@@ -225,8 +227,6 @@
struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node,
struct rsnd_mod *mod, char *name);
-#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
-
/*
* R-Car sound mod
*/
@@ -330,8 +330,9 @@
#define rsnd_mod_to_priv(mod) ((mod)->priv)
#define rsnd_mod_to_dma(mod) (&(mod)->dma)
#define rsnd_mod_id(mod) ((mod) ? (mod)->id : -1)
-#define rsnd_mod_hw_start(mod) clk_enable((mod)->clk)
-#define rsnd_mod_hw_stop(mod) clk_disable((mod)->clk)
+#define rsnd_mod_power_on(mod) clk_enable((mod)->clk)
+#define rsnd_mod_power_off(mod) clk_disable((mod)->clk)
+#define rsnd_mod_get(ip) (&(ip)->mod)
int rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
@@ -571,9 +572,12 @@
void rsnd_ssi_remove(struct platform_device *pdev,
struct rsnd_priv *priv);
struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
-int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod);
-int rsnd_ssi_use_busif(struct rsnd_dai_stream *io, struct rsnd_mod *mod);
+int rsnd_ssi_use_busif(struct rsnd_dai_stream *io);
+
+#define rsnd_ssi_is_pin_sharing(io) \
+ __rsnd_ssi_is_pin_sharing(rsnd_io_to_mod_ssi(io))
+int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
/*
* R-Car SRC
@@ -627,4 +631,15 @@
struct rsnd_priv *priv);
struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id);
+#ifdef DEBUG
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type);
+#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI)
+#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC)
+#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC)
+#else
+#define rsnd_mod_confirm_ssi(mssi)
+#define rsnd_mod_confirm_src(msrc)
+#define rsnd_mod_confirm_dvc(mdvc)
+#endif
+
#endif
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 89a18e1..261b502 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -159,7 +159,7 @@
/*
* SSI_MODE1
*/
- if (rsnd_ssi_is_pin_sharing(ssi_mod)) {
+ if (rsnd_ssi_is_pin_sharing(io)) {
int shift = -1;
switch (ssi_id) {
case 1:
@@ -352,7 +352,7 @@
{
struct rsnd_src *src = rsnd_mod_to_src(mod);
- rsnd_mod_hw_start(mod);
+ rsnd_mod_power_on(mod);
rsnd_src_soft_reset(mod);
@@ -373,7 +373,7 @@
struct rsnd_src *src = rsnd_mod_to_src(mod);
struct device *dev = rsnd_priv_to_dev(priv);
- rsnd_mod_hw_stop(mod);
+ rsnd_mod_power_off(mod);
if (src->err)
dev_warn(dev, "%s[%d] under/over flow err = %d\n",
@@ -918,11 +918,10 @@
rsnd_mod_write(mod, SRC_IFSVR, fsrate);
}
-static int rsnd_src_pcm_new(struct rsnd_mod *mod,
+static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct snd_soc_pcm_runtime *rtd)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct rsnd_src *src = rsnd_mod_to_src(mod);
int ret;
@@ -932,12 +931,6 @@
*/
/*
- * Gen1 is not supported
- */
- if (rsnd_is_gen1(priv))
- return 0;
-
- /*
* SRC sync convert needs clock master
*/
if (!rsnd_rdai_is_clk_master(rdai))
@@ -975,7 +968,7 @@
.start = rsnd_src_start_gen2,
.stop = rsnd_src_stop_gen2,
.hw_params = rsnd_src_hw_params,
- .pcm_new = rsnd_src_pcm_new,
+ .pcm_new = rsnd_src_pcm_new_gen2,
};
struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
@@ -983,7 +976,7 @@
if (WARN_ON(id < 0 || id >= rsnd_src_nr(priv)))
id = 0;
- return &((struct rsnd_src *)(priv->src) + id)->mod;
+ return rsnd_mod_get((struct rsnd_src *)(priv->src) + id);
}
static void rsnd_of_parse_src(struct platform_device *pdev,
@@ -1043,8 +1036,10 @@
int i, nr, ret;
ops = NULL;
- if (rsnd_is_gen1(priv))
+ if (rsnd_is_gen1(priv)) {
ops = &rsnd_src_gen1_ops;
+ dev_warn(dev, "Gen1 support will be removed soon\n");
+ }
if (rsnd_is_gen2(priv))
ops = &rsnd_src_gen2_ops;
if (!ops) {
@@ -1078,7 +1073,7 @@
src->info = &info->src_info[i];
- ret = rsnd_mod_init(priv, &src->mod, ops, clk, RSND_MOD_SRC, i);
+ ret = rsnd_mod_init(priv, rsnd_mod_get(src), ops, clk, RSND_MOD_SRC, i);
if (ret)
return ret;
}
@@ -1093,6 +1088,6 @@
int i;
for_each_rsnd_src(src, priv, i) {
- rsnd_mod_quit(&src->mod);
+ rsnd_mod_quit(rsnd_mod_get(src));
}
}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index d45b9a7..1427ec2 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -79,7 +79,6 @@
#define rsnd_ssi_nr(priv) ((priv)->ssi_nr)
#define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
-#define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma))
#define rsnd_ssi_pio_available(ssi) ((ssi)->info->irq > 0)
#define rsnd_ssi_parent(ssi) ((ssi)->parent)
#define rsnd_ssi_mode_flags(p) ((p)->info->flags)
@@ -87,8 +86,9 @@
#define rsnd_ssi_of_node(priv) \
of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,ssi")
-int rsnd_ssi_use_busif(struct rsnd_dai_stream *io, struct rsnd_mod *mod)
+int rsnd_ssi_use_busif(struct rsnd_dai_stream *io)
{
+ struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io);
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
int use_busif = 0;
@@ -128,10 +128,8 @@
struct rsnd_priv *priv = rsnd_io_to_priv(io);
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
struct device *dev = rsnd_priv_to_dev(priv);
- int i, j, ret;
- int adg_clk_div_table[] = {
- 1, 6, /* see adg.c */
- };
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+ int j, ret;
int ssi_clk_mul_table[] = {
1, 2, 4, 8, 16, 6, 12,
};
@@ -141,28 +139,25 @@
/*
* Find best clock, and try to start ADG
*/
- for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) {
- for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
+ for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
- /*
- * this driver is assuming that
- * system word is 64fs (= 2 x 32bit)
- * see rsnd_ssi_init()
- */
- main_rate = rate / adg_clk_div_table[i]
- * 32 * 2 * ssi_clk_mul_table[j];
+ /*
+ * this driver is assuming that
+ * system word is 64fs (= 2 x 32bit)
+ * see rsnd_ssi_init()
+ */
+ main_rate = rate * 32 * 2 * ssi_clk_mul_table[j];
- ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate);
- if (0 == ret) {
- ssi->cr_clk = FORCE | SWL_32 |
- SCKD | SWSD | CKDV(j);
+ ret = rsnd_adg_ssi_clk_try_start(mod, main_rate);
+ if (0 == ret) {
+ ssi->cr_clk = FORCE | SWL_32 |
+ SCKD | SWSD | CKDV(j);
- dev_dbg(dev, "%s[%d] outputs %u Hz\n",
- rsnd_mod_name(&ssi->mod),
- rsnd_mod_id(&ssi->mod), rate);
+ dev_dbg(dev, "%s[%d] outputs %u Hz\n",
+ rsnd_mod_name(mod),
+ rsnd_mod_id(mod), rate);
- return 0;
- }
+ return 0;
}
}
@@ -172,8 +167,10 @@
static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi)
{
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
ssi->cr_clk = 0;
- rsnd_adg_ssi_clk_stop(&ssi->mod);
+ rsnd_adg_ssi_clk_stop(mod);
}
static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
@@ -182,11 +179,12 @@
struct rsnd_priv *priv = rsnd_io_to_priv(io);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
u32 cr_mode;
u32 cr;
if (0 == ssi->usrcnt) {
- rsnd_mod_hw_start(&ssi->mod);
+ rsnd_mod_power_on(mod);
if (rsnd_rdai_is_clk_master(rdai)) {
struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -198,7 +196,7 @@
}
}
- if (rsnd_ssi_is_dma_mode(&ssi->mod)) {
+ if (rsnd_ssi_is_dma_mode(mod)) {
cr_mode = UIEN | OIEN | /* over/under run */
DMEN; /* DMA : enable DMA */
} else {
@@ -210,24 +208,25 @@
cr_mode |
EN;
- rsnd_mod_write(&ssi->mod, SSICR, cr);
+ rsnd_mod_write(mod, SSICR, cr);
/* enable WS continue */
if (rsnd_rdai_is_clk_master(rdai))
- rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
+ rsnd_mod_write(mod, SSIWSR, CONT);
/* clear error status */
- rsnd_mod_write(&ssi->mod, SSISR, 0);
+ rsnd_mod_write(mod, SSISR, 0);
ssi->usrcnt++;
dev_dbg(dev, "%s[%d] hw started\n",
- rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
}
static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
struct device *dev = rsnd_priv_to_dev(priv);
u32 cr;
@@ -247,15 +246,15 @@
cr = ssi->cr_own |
ssi->cr_clk;
- rsnd_mod_write(&ssi->mod, SSICR, cr | EN);
- rsnd_ssi_status_check(&ssi->mod, DIRQ);
+ rsnd_mod_write(mod, SSICR, cr | EN);
+ rsnd_ssi_status_check(mod, DIRQ);
/*
* disable SSI,
* and, wait idle state
*/
- rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */
- rsnd_ssi_status_check(&ssi->mod, IIRQ);
+ rsnd_mod_write(mod, SSICR, cr); /* disabled all */
+ rsnd_ssi_status_check(mod, IIRQ);
if (rsnd_rdai_is_clk_master(rdai)) {
struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -266,13 +265,13 @@
rsnd_ssi_master_clk_stop(ssi);
}
- rsnd_mod_hw_stop(&ssi->mod);
+ rsnd_mod_power_off(mod);
ssi->chan = 0;
}
dev_dbg(dev, "%s[%d] hw stopped\n",
- rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
}
/*
@@ -371,7 +370,7 @@
/* It will be removed on rsnd_ssi_hw_stop */
ssi->chan = chan;
if (ssi_parent)
- return rsnd_ssi_hw_params(&ssi_parent->mod, io,
+ return rsnd_ssi_hw_params(rsnd_mod_get(ssi_parent), io,
substream, params);
return 0;
@@ -379,12 +378,14 @@
static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status)
{
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
/* under/over flow error */
if (status & (UIRQ | OIRQ)) {
ssi->err++;
/* clear error status */
- rsnd_mod_write(&ssi->mod, SSISR, 0);
+ rsnd_mod_write(mod, SSISR, 0);
}
}
@@ -394,7 +395,7 @@
{
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
- rsnd_src_ssiu_start(mod, io, rsnd_ssi_use_busif(io, mod));
+ rsnd_src_ssiu_start(mod, io, rsnd_ssi_use_busif(io));
rsnd_ssi_hw_start(ssi, io);
@@ -554,7 +555,7 @@
rsnd_dma_quit(io, rsnd_mod_to_dma(mod));
/* PIO will request IRQ again */
- devm_free_irq(dev, irq, ssi);
+ devm_free_irq(dev, irq, mod);
return 0;
}
@@ -613,7 +614,7 @@
int is_play = rsnd_io_is_play(io);
char *name;
- if (rsnd_ssi_use_busif(io, mod))
+ if (rsnd_ssi_use_busif(io))
name = is_play ? "rxu" : "txu";
else
name = is_play ? "rx" : "tx";
@@ -656,10 +657,10 @@
if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv)))
id = 0;
- return &((struct rsnd_ssi *)(priv->ssi) + id)->mod;
+ return rsnd_mod_get((struct rsnd_ssi *)(priv->ssi) + id);
}
-int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
+int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
{
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
@@ -668,10 +669,12 @@
static void rsnd_ssi_parent_setup(struct rsnd_priv *priv, struct rsnd_ssi *ssi)
{
- if (!rsnd_ssi_is_pin_sharing(&ssi->mod))
+ struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
+ if (!__rsnd_ssi_is_pin_sharing(mod))
return;
- switch (rsnd_mod_id(&ssi->mod)) {
+ switch (rsnd_mod_id(mod)) {
case 1:
case 2:
ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 0));
@@ -697,9 +700,6 @@
struct device *dev = &pdev->dev;
int nr, i;
- if (!of_data)
- return;
-
node = rsnd_ssi_of_node(priv);
if (!node)
return;
@@ -794,7 +794,8 @@
else if (rsnd_ssi_pio_available(ssi))
ops = &rsnd_ssi_pio_ops;
- ret = rsnd_mod_init(priv, &ssi->mod, ops, clk, RSND_MOD_SSI, i);
+ ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk,
+ RSND_MOD_SSI, i);
if (ret)
return ret;
@@ -811,6 +812,6 @@
int i;
for_each_rsnd_ssi(ssi, priv, i) {
- rsnd_mod_quit(&ssi->mod);
+ rsnd_mod_quit(rsnd_mod_get(ssi));
}
}
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index abb0d95..76b2ab8 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -738,7 +738,7 @@
struct siu_info *info;
int ret;
- info = kmalloc(sizeof(*info), GFP_KERNEL);
+ info = devm_kmalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
if (!info)
return -ENOMEM;
siu_i2s_data = info;
@@ -746,7 +746,7 @@
ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev);
if (ret)
- goto ereqfw;
+ return ret;
/*
* Loaded firmware is "const" - read only, but we have to modify it in
@@ -757,89 +757,52 @@
release_firmware(fw_entry);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- ret = -ENODEV;
- goto egetres;
- }
+ if (!res)
+ return -ENODEV;
- region = request_mem_region(res->start, resource_size(res),
- pdev->name);
+ region = devm_request_mem_region(&pdev->dev, res->start,
+ resource_size(res), pdev->name);
if (!region) {
dev_err(&pdev->dev, "SIU region already claimed\n");
- ret = -EBUSY;
- goto ereqmemreg;
+ return -EBUSY;
}
- ret = -ENOMEM;
- info->pram = ioremap(res->start, PRAM_SIZE);
+ info->pram = devm_ioremap(&pdev->dev, res->start, PRAM_SIZE);
if (!info->pram)
- goto emappram;
- info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE);
+ return -ENOMEM;
+ info->xram = devm_ioremap(&pdev->dev, res->start + XRAM_OFFSET,
+ XRAM_SIZE);
if (!info->xram)
- goto emapxram;
- info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE);
+ return -ENOMEM;
+ info->yram = devm_ioremap(&pdev->dev, res->start + YRAM_OFFSET,
+ YRAM_SIZE);
if (!info->yram)
- goto emapyram;
- info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) -
- REG_OFFSET);
+ return -ENOMEM;
+ info->reg = devm_ioremap(&pdev->dev, res->start + REG_OFFSET,
+ resource_size(res) - REG_OFFSET);
if (!info->reg)
- goto emapreg;
+ return -ENOMEM;
dev_set_drvdata(&pdev->dev, info);
/* register using ARRAY version so we can keep dai name */
- ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component,
- &siu_i2s_dai, 1);
+ ret = devm_snd_soc_register_component(&pdev->dev, &siu_i2s_component,
+ &siu_i2s_dai, 1);
if (ret < 0)
- goto edaiinit;
+ return ret;
- ret = snd_soc_register_platform(&pdev->dev, &siu_platform);
+ ret = devm_snd_soc_register_platform(&pdev->dev, &siu_platform);
if (ret < 0)
- goto esocregp;
+ return ret;
pm_runtime_enable(&pdev->dev);
- return ret;
-
-esocregp:
- snd_soc_unregister_component(&pdev->dev);
-edaiinit:
- iounmap(info->reg);
-emapreg:
- iounmap(info->yram);
-emapyram:
- iounmap(info->xram);
-emapxram:
- iounmap(info->pram);
-emappram:
- release_mem_region(res->start, resource_size(res));
-ereqmemreg:
-egetres:
-ereqfw:
- kfree(info);
-
- return ret;
+ return 0;
}
static int siu_remove(struct platform_device *pdev)
{
- struct siu_info *info = dev_get_drvdata(&pdev->dev);
- struct resource *res;
-
pm_runtime_disable(&pdev->dev);
-
- snd_soc_unregister_platform(&pdev->dev);
- snd_soc_unregister_component(&pdev->dev);
-
- iounmap(info->reg);
- iounmap(info->yram);
- iounmap(info->xram);
- iounmap(info->pram);
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (res)
- release_mem_region(res->start, resource_size(res));
- kfree(info);
-
return 0;
}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 025c38f..12a9820 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -612,8 +612,15 @@
.get_codec_caps = soc_compr_get_codec_caps
};
-/* create a new compress */
-int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
+/**
+ * snd_soc_new_compress - create a new compress.
+ *
+ * @rtd: The runtime for which we will create compress
+ * @num: the device index number (zero based - shared with normal PCMs)
+ *
+ * Return: 0 for success, else error.
+ */
+int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_platform *platform = rtd->platform;
@@ -703,3 +710,4 @@
kfree(compr);
return ret;
}
+EXPORT_SYMBOL_GPL(snd_soc_new_compress);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6173d15..24b0960 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1370,9 +1370,9 @@
soc_dpcm_debugfs_add(rtd);
#endif
- if (cpu_dai->driver->compress_dai) {
+ if (cpu_dai->driver->compress_new) {
/*create compress_device"*/
- ret = soc_new_compress(rtd, num);
+ ret = cpu_dai->driver->compress_new(rtd, num);
if (ret < 0) {
dev_err(card->dev, "ASoC: can't create compress %s\n",
dai_link->stream_name);
@@ -3291,13 +3291,38 @@
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
+static int snd_soc_of_get_slot_mask(struct device_node *np,
+ const char *prop_name,
+ unsigned int *mask)
+{
+ u32 val;
+ const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
+ int i;
+
+ if (!of_slot_mask)
+ return 0;
+ val /= sizeof(u32);
+ for (i = 0; i < val; i++)
+ if (be32_to_cpup(&of_slot_mask[i]))
+ *mask |= (1 << i);
+
+ return val;
+}
+
int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *tx_mask,
+ unsigned int *rx_mask,
unsigned int *slots,
unsigned int *slot_width)
{
u32 val;
int ret;
+ if (tx_mask)
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask);
+ if (rx_mask)
+ snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask);
+
if (of_property_read_bool(np, "dai-tdm-slot-num")) {
ret = of_property_read_u32(np, "dai-tdm-slot-num", &val);
if (ret)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index ff8bda4..016eba1 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -509,6 +509,18 @@
}
/**
+ * snd_soc_dapm_kcontrol_widget() - Returns the widget associated to a
+ * kcontrol
+ * @kcontrol: The kcontrol
+ */
+struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget(
+ struct snd_kcontrol *kcontrol)
+{
+ return dapm_kcontrol_get_wlist(kcontrol)->widgets[0];
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_widget);
+
+/**
* snd_soc_dapm_kcontrol_dapm() - Returns the dapm context associated to a
* kcontrol
* @kcontrol: The kcontrol
@@ -779,7 +791,7 @@
* Determine if a kcontrol is shared. If it is, look it up. If it isn't,
* create it. Either way, add the widget into the control's widget list
*/
-static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
+static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w,
int kci)
{
struct snd_soc_dapm_context *dapm = w->dapm;
@@ -810,6 +822,7 @@
switch (w->id) {
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer:
+ case snd_soc_dapm_pga:
wname_in_long_name = true;
kcname_in_long_name = true;
break;
@@ -899,7 +912,7 @@
continue;
if (!w->kcontrols[i]) {
- ret = dapm_create_or_share_mixmux_kcontrol(w, i);
+ ret = dapm_create_or_share_kcontrol(w, i);
if (ret < 0)
return ret;
}
@@ -952,7 +965,7 @@
return -EINVAL;
}
- ret = dapm_create_or_share_mixmux_kcontrol(w, 0);
+ ret = dapm_create_or_share_kcontrol(w, 0);
if (ret < 0)
return ret;
@@ -967,9 +980,13 @@
/* create new dapm volume control */
static int dapm_new_pga(struct snd_soc_dapm_widget *w)
{
- if (w->num_kcontrols)
- dev_err(w->dapm->dev,
- "ASoC: PGA controls not supported: '%s'\n", w->name);
+ int i, ret;
+
+ for (i = 0; i < w->num_kcontrols; i++) {
+ ret = dapm_create_or_share_kcontrol(w, i);
+ if (ret < 0)
+ return ret;
+ }
return 0;
}
@@ -3473,11 +3490,29 @@
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+ if (source->driver->ops && source->driver->ops->startup) {
+ ret = source->driver->ops->startup(&substream, source);
+ if (ret < 0) {
+ dev_err(source->dev,
+ "ASoC: startup() failed: %d\n", ret);
+ goto out;
+ }
+ source->active++;
+ }
ret = soc_dai_hw_params(&substream, params, source);
if (ret < 0)
goto out;
substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+ if (sink->driver->ops && sink->driver->ops->startup) {
+ ret = sink->driver->ops->startup(&substream, sink);
+ if (ret < 0) {
+ dev_err(sink->dev,
+ "ASoC: startup() failed: %d\n", ret);
+ goto out;
+ }
+ sink->active++;
+ }
ret = soc_dai_hw_params(&substream, params, sink);
if (ret < 0)
goto out;
@@ -3497,6 +3532,18 @@
if (ret != 0 && ret != -ENOTSUPP)
dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret);
ret = 0;
+
+ source->active--;
+ if (source->driver->ops && source->driver->ops->shutdown) {
+ substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+ source->driver->ops->shutdown(&substream, source);
+ }
+
+ sink->active--;
+ if (sink->driver->ops && sink->driver->ops->shutdown) {
+ substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+ sink->driver->ops->shutdown(&substream, sink);
+ }
break;
default:
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 05977ae..ecd38e5 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -588,16 +588,16 @@
/**
* snd_soc_limit_volume - Set new limit to an existing volume control.
*
- * @codec: where to look for the control
+ * @card: where to look for the control
* @name: Name of the control
* @max: new maximum limit
*
* Return 0 for success, else error.
*/
-int snd_soc_limit_volume(struct snd_soc_codec *codec,
+int snd_soc_limit_volume(struct snd_soc_card *card,
const char *name, int max)
{
- struct snd_card *card = codec->component.card->snd_card;
+ struct snd_card *snd_card = card->snd_card;
struct snd_kcontrol *kctl;
struct soc_mixer_control *mc;
int found = 0;
@@ -607,7 +607,7 @@
if (unlikely(!name || max <= 0))
return -EINVAL;
- list_for_each_entry(kctl, &card->controls, list) {
+ list_for_each_entry(kctl, &snd_card->controls, list) {
if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) {
found = 1;
break;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 70e4b9d..c86dc96 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -34,6 +34,24 @@
#define DPCM_MAX_BE_USERS 8
+/*
+ * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
+ *
+ * Returns true if the DAI supports the indicated stream type.
+ */
+static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_pcm_stream *codec_stream;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_stream = &dai->driver->playback;
+ else
+ codec_stream = &dai->driver->capture;
+
+ /* If the codec specifies any rate at all, it supports the stream. */
+ return codec_stream->rates;
+}
+
/**
* snd_soc_runtime_activate() - Increment active count for PCM runtime components
* @rtd: ASoC PCM runtime that is activated
@@ -182,9 +200,9 @@
dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n",
soc_dai->rate);
- ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ ret = snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
- soc_dai->rate, soc_dai->rate);
+ soc_dai->rate);
if (ret < 0) {
dev_err(soc_dai->dev,
"ASoC: Unable to apply rate constraint: %d\n",
@@ -198,9 +216,8 @@
dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d channel(s)\n",
soc_dai->channels);
- ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ ret = snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
- soc_dai->channels,
soc_dai->channels);
if (ret < 0) {
dev_err(soc_dai->dev,
@@ -215,9 +232,8 @@
dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d sample bits\n",
soc_dai->sample_bits);
- ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ ret = snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- soc_dai->sample_bits,
soc_dai->sample_bits);
if (ret < 0) {
dev_err(soc_dai->dev,
@@ -371,6 +387,20 @@
/* first calculate min/max only for CODECs in the DAI link */
for (i = 0; i < rtd->num_codecs; i++) {
+
+ /*
+ * Skip CODECs which don't support the current stream type.
+ * Otherwise, since the rate, channel, and format values will
+ * zero in that case, we would have no usable settings left,
+ * causing the resulting setup to fail.
+ * At least one CODEC should match, otherwise we should have
+ * bailed out on a higher level, since there would be no
+ * CODEC to support the transfer direction in that case.
+ */
+ if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+ substream->stream))
+ continue;
+
codec_dai_drv = rtd->codec_dais[i]->driver;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
codec_stream = &codec_dai_drv->playback;
@@ -827,6 +857,23 @@
struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
struct snd_pcm_hw_params codec_params;
+ /*
+ * Skip CODECs which don't support the current stream type,
+ * the idea being that if a CODEC is not used for the currently
+ * set up transfer direction, it should not need to be
+ * configured, especially since the configuration used might
+ * not even be supported by that CODEC. There may be cases
+ * however where a CODEC needs to be set up although it is
+ * actually not being used for the transfer, e.g. if a
+ * capture-only CODEC is acting as an LRCLK and/or BCLK master
+ * for the DAI link including a playback-only CODEC.
+ * If this becomes necessary, we will have to augment the
+ * machine driver setup with information on how to act, so
+ * we can do the right thing here.
+ */
+ if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+ continue;
+
/* copy params for each codec */
codec_params = *params;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 69d01cd..8d7ec80 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1558,7 +1558,7 @@
pcm_dai = (struct snd_soc_tplg_pcm_dai *)tplg->pos;
if (soc_tplg_check_elem_count(tplg,
- sizeof(struct snd_soc_tplg_pcm_dai), count,
+ sizeof(struct snd_soc_tplg_pcm), count,
hdr->payload_size, "PCM DAI")) {
dev_err(tplg->dev, "ASoC: invalid count %d for PCM DAI elems\n",
count);
@@ -1566,7 +1566,7 @@
}
dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count);
- tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count;
+ tplg->pos += sizeof(struct snd_soc_tplg_pcm) * count;
dobj = kzalloc(sizeof(struct snd_soc_dobj), GFP_KERNEL);
if (dobj == NULL)
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
new file mode 100644
index 0000000..84c72ec
--- /dev/null
+++ b/sound/soc/sunxi/Kconfig
@@ -0,0 +1,11 @@
+menu "Allwinner SoC Audio support"
+
+config SND_SUN4I_CODEC
+ tristate "Allwinner A10 Codec Support"
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ select REGMAP_MMIO
+ help
+ Select Y or M to add support for the Codec embedded in the Allwinner
+ A10 and affiliated SoCs.
+
+endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
new file mode 100644
index 0000000..ea8a08c
--- /dev/null
+++ b/sound/soc/sunxi/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
+
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
new file mode 100644
index 0000000..bcbf4da
--- /dev/null
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -0,0 +1,712 @@
+/*
+ * Copyright 2014 Emilio López <emilio@elopez.com.ar>
+ * Copyright 2014 Jon Smirl <jonsmirl@gmail.com>
+ * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com>
+ *
+ * Based on the Allwinner SDK driver, released under the GPL.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_address.h>
+#include <linux/clk.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/dmaengine_pcm.h>
+
+/* Codec DAC register offsets and bit fields */
+#define SUN4I_CODEC_DAC_DPC (0x00)
+#define SUN4I_CODEC_DAC_DPC_EN_DA (31)
+#define SUN4I_CODEC_DAC_DPC_DVOL (12)
+#define SUN4I_CODEC_DAC_FIFOC (0x04)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_FS (29)
+#define SUN4I_CODEC_DAC_FIFOC_FIR_VERSION (28)
+#define SUN4I_CODEC_DAC_FIFOC_SEND_LASAT (26)
+#define SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE (24)
+#define SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT (21)
+#define SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL (8)
+#define SUN4I_CODEC_DAC_FIFOC_MONO_EN (6)
+#define SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS (5)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN (4)
+#define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH (0)
+#define SUN4I_CODEC_DAC_FIFOS (0x08)
+#define SUN4I_CODEC_DAC_TXDATA (0x0c)
+#define SUN4I_CODEC_DAC_ACTL (0x10)
+#define SUN4I_CODEC_DAC_ACTL_DACAENR (31)
+#define SUN4I_CODEC_DAC_ACTL_DACAENL (30)
+#define SUN4I_CODEC_DAC_ACTL_MIXEN (29)
+#define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15)
+#define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14)
+#define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13)
+#define SUN4I_CODEC_DAC_ACTL_DACPAS (8)
+#define SUN4I_CODEC_DAC_ACTL_MIXPAS (7)
+#define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6)
+#define SUN4I_CODEC_DAC_ACTL_PA_VOL (0)
+#define SUN4I_CODEC_DAC_TUNE (0x14)
+#define SUN4I_CODEC_DAC_DEBUG (0x18)
+
+/* Codec ADC register offsets and bit fields */
+#define SUN4I_CODEC_ADC_FIFOC (0x1c)
+#define SUN4I_CODEC_ADC_FIFOC_EN_AD (28)
+#define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24)
+#define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8)
+#define SUN4I_CODEC_ADC_FIFOC_MONO_EN (7)
+#define SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS (6)
+#define SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN (4)
+#define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH (0)
+#define SUN4I_CODEC_ADC_FIFOS (0x20)
+#define SUN4I_CODEC_ADC_RXDATA (0x24)
+#define SUN4I_CODEC_ADC_ACTL (0x28)
+#define SUN4I_CODEC_ADC_ACTL_ADC_R_EN (31)
+#define SUN4I_CODEC_ADC_ACTL_ADC_L_EN (30)
+#define SUN4I_CODEC_ADC_ACTL_PREG1EN (29)
+#define SUN4I_CODEC_ADC_ACTL_PREG2EN (28)
+#define SUN4I_CODEC_ADC_ACTL_VMICEN (27)
+#define SUN4I_CODEC_ADC_ACTL_VADCG (20)
+#define SUN4I_CODEC_ADC_ACTL_ADCIS (17)
+#define SUN4I_CODEC_ADC_ACTL_PA_EN (4)
+#define SUN4I_CODEC_ADC_ACTL_DDE (3)
+#define SUN4I_CODEC_ADC_DEBUG (0x2c)
+
+/* Other various ADC registers */
+#define SUN4I_CODEC_DAC_TXCNT (0x30)
+#define SUN4I_CODEC_ADC_RXCNT (0x34)
+#define SUN4I_CODEC_AC_SYS_VERI (0x38)
+#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c)
+
+struct sun4i_codec {
+ struct device *dev;
+ struct regmap *regmap;
+ struct clk *clk_apb;
+ struct clk *clk_module;
+
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+};
+
+static void sun4i_codec_start_playback(struct sun4i_codec *scodec)
+{
+ /*
+ * FIXME: according to the BSP, we might need to drive a PA
+ * GPIO high here on some boards
+ */
+
+ /* Flush TX FIFO */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+ /* Enable DAC DRQ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN));
+}
+
+static void sun4i_codec_stop_playback(struct sun4i_codec *scodec)
+{
+ /*
+ * FIXME: according to the BSP, we might need to drive a PA
+ * GPIO low here on some boards
+ */
+
+ /* Disable DAC DRQ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+ 0);
+}
+
+static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ sun4i_codec_start_playback(scodec);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sun4i_codec_stop_playback(scodec);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int sun4i_codec_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+ u32 val;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ /* Flush the TX FIFO */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+ /* Set TX FIFO Empty Trigger Level */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 0x3f << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL,
+ 0xf << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL);
+
+ if (substream->runtime->rate > 32000)
+ /* Use 64 bits FIR filter */
+ val = 0;
+ else
+ /* Use 32 bits FIR filter */
+ val = BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION);
+
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION),
+ val);
+
+ /* Send zeros when we have an underrun */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT),
+ 0);
+
+ return 0;
+}
+
+static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params)
+{
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 176400:
+ case 88200:
+ case 44100:
+ case 33075:
+ case 22050:
+ case 14700:
+ case 11025:
+ case 7350:
+ return 22579200;
+
+ case 192000:
+ case 96000:
+ case 48000:
+ case 32000:
+ case 24000:
+ case 16000:
+ case 12000:
+ case 8000:
+ return 24576000;
+
+ default:
+ return 0;
+ }
+}
+
+static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params)
+{
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 192000:
+ case 176400:
+ return 6;
+
+ case 96000:
+ case 88200:
+ return 7;
+
+ case 48000:
+ case 44100:
+ return 0;
+
+ case 32000:
+ case 33075:
+ return 1;
+
+ case 24000:
+ case 22050:
+ return 2;
+
+ case 16000:
+ case 14700:
+ return 3;
+
+ case 12000:
+ case 11025:
+ return 4;
+
+ case 8000:
+ case 7350:
+ return 5;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int sun4i_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+ unsigned long clk_freq;
+ int ret, hwrate;
+ u32 val;
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENOTSUPP;
+
+ clk_freq = sun4i_codec_get_mod_freq(params);
+ if (!clk_freq)
+ return -EINVAL;
+
+ ret = clk_set_rate(scodec->clk_module, clk_freq);
+ if (ret)
+ return ret;
+
+ hwrate = sun4i_codec_get_hw_rate(params);
+ if (hwrate < 0)
+ return hwrate;
+
+ /* Set DAC sample rate */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 7 << SUN4I_CODEC_DAC_FIFOC_DAC_FS,
+ hwrate << SUN4I_CODEC_DAC_FIFOC_DAC_FS);
+
+ /* Set the number of channels we want to use */
+ if (params_channels(params) == 1)
+ val = BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN);
+ else
+ val = 0;
+
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN),
+ val);
+
+ /* Set the number of sample bits to either 16 or 24 bits */
+ if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) {
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS));
+
+ /* Set TX FIFO mode to padding the LSBs with 0 */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+ 0);
+
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ } else {
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+ 0);
+
+ /* Set TX FIFO mode to repeat the MSB */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+ BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE));
+
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ }
+
+ return 0;
+}
+
+static int sun4i_codec_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ /*
+ * Stop issuing DRQ when we have room for less than 16 samples
+ * in our TX FIFO
+ */
+ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+ 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT,
+ 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT);
+
+ return clk_prepare_enable(scodec->clk_module);
+}
+
+static void sun4i_codec_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+ clk_disable_unprepare(scodec->clk_module);
+}
+
+static const struct snd_soc_dai_ops sun4i_codec_dai_ops = {
+ .startup = sun4i_codec_startup,
+ .shutdown = sun4i_codec_shutdown,
+ .trigger = sun4i_codec_trigger,
+ .hw_params = sun4i_codec_hw_params,
+ .prepare = sun4i_codec_prepare,
+};
+
+static struct snd_soc_dai_driver sun4i_codec_dai = {
+ .name = "Codec",
+ .ops = &sun4i_codec_dai_ops,
+ .playback = {
+ .stream_name = "Codec Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .rates = SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 24,
+ },
+};
+
+/*** Codec ***/
+static const struct snd_kcontrol_new sun4i_codec_pa_mute =
+ SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0);
+
+static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1);
+
+static const struct snd_kcontrol_new sun4i_codec_widgets[] = {
+ SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0,
+ sun4i_codec_pa_volume_scale),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACPAS, 1, 0),
+ SOC_DAPM_SINGLE("Mixer Playback Switch", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = {
+ /* Digital parts of the DACs */
+ SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC,
+ SUN4I_CODEC_DAC_DPC_EN_DA, 0,
+ NULL, 0),
+
+ /* Analog parts of the DACs */
+ SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACAENL, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_DACAENR, 0),
+
+ /* Mixers */
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ sun4i_codec_left_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ sun4i_codec_right_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_right_mixer_controls)),
+
+ /* Global Mixer Enable */
+ SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL,
+ SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0),
+
+ /* Pre-Amplifier */
+ SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL,
+ SUN4I_CODEC_ADC_ACTL_PA_EN, 0,
+ sun4i_codec_pa_mixer_controls,
+ ARRAY_SIZE(sun4i_codec_pa_mixer_controls)),
+ SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0,
+ &sun4i_codec_pa_mute),
+
+ SND_SOC_DAPM_OUTPUT("HP Right"),
+ SND_SOC_DAPM_OUTPUT("HP Left"),
+};
+
+static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = {
+ /* Left DAC Routes */
+ { "Left DAC", NULL, "DAC" },
+
+ /* Right DAC Routes */
+ { "Right DAC", NULL, "DAC" },
+
+ /* Right Mixer Routes */
+ { "Right Mixer", NULL, "Mixer Enable" },
+ { "Right Mixer", "Left DAC Playback Switch", "Left DAC" },
+ { "Right Mixer", "Right DAC Playback Switch", "Right DAC" },
+
+ /* Left Mixer Routes */
+ { "Left Mixer", NULL, "Mixer Enable" },
+ { "Left Mixer", "Left DAC Playback Switch", "Left DAC" },
+
+ /* Pre-Amplifier Mixer Routes */
+ { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" },
+ { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" },
+ { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" },
+ { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" },
+
+ /* PA -> HP path */
+ { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" },
+ { "HP Right", NULL, "Pre-Amplifier Mute" },
+ { "HP Left", NULL, "Pre-Amplifier Mute" },
+};
+
+static struct snd_soc_codec_driver sun4i_codec_codec = {
+ .controls = sun4i_codec_widgets,
+ .num_controls = ARRAY_SIZE(sun4i_codec_widgets),
+ .dapm_widgets = sun4i_codec_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets),
+ .dapm_routes = sun4i_codec_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes),
+};
+
+static const struct snd_soc_component_driver sun4i_codec_component = {
+ .name = "sun4i-codec",
+};
+
+#define SUN4I_CODEC_RATES SNDRV_PCM_RATE_8000_192000
+#define SUN4I_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static int sun4i_codec_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai);
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+ snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data,
+ NULL);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver dummy_cpu_dai = {
+ .name = "sun4i-codec-cpu-dai",
+ .probe = sun4i_codec_dai_probe,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SUN4I_CODEC_RATES,
+ .formats = SUN4I_CODEC_FORMATS,
+ .sig_bits = 24,
+ },
+};
+
+static const struct regmap_config sun4i_codec_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL,
+};
+
+static const struct of_device_id sun4i_codec_of_match[] = {
+ { .compatible = "allwinner,sun4i-a10-codec" },
+ { .compatible = "allwinner,sun7i-a20-codec" },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sun4i_codec_of_match);
+
+static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev,
+ int *num_links)
+{
+ struct snd_soc_dai_link *link = devm_kzalloc(dev, sizeof(*link),
+ GFP_KERNEL);
+ if (!link)
+ return NULL;
+
+ link->name = "cdc";
+ link->stream_name = "CDC PCM";
+ link->codec_dai_name = "Codec";
+ link->cpu_dai_name = dev_name(dev);
+ link->codec_name = dev_name(dev);
+ link->platform_name = dev_name(dev);
+ link->dai_fmt = SND_SOC_DAIFMT_I2S;
+
+ *num_links = 1;
+
+ return link;
+};
+
+static struct snd_soc_card *sun4i_codec_create_card(struct device *dev)
+{
+ struct snd_soc_card *card;
+
+ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return NULL;
+
+ card->dai_link = sun4i_codec_create_link(dev, &card->num_links);
+ if (!card->dai_link)
+ return NULL;
+
+ card->dev = dev;
+ card->name = "sun4i-codec";
+
+ return card;
+};
+
+static int sun4i_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card;
+ struct sun4i_codec *scodec;
+ struct resource *res;
+ void __iomem *base;
+ int ret;
+
+ scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL);
+ if (!scodec)
+ return -ENOMEM;
+
+ scodec->dev = &pdev->dev;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(base)) {
+ dev_err(&pdev->dev, "Failed to map the registers\n");
+ return PTR_ERR(base);
+ }
+
+ scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sun4i_codec_regmap_config);
+ if (IS_ERR(scodec->regmap)) {
+ dev_err(&pdev->dev, "Failed to create our regmap\n");
+ return PTR_ERR(scodec->regmap);
+ }
+
+ /* Get the clocks from the DT */
+ scodec->clk_apb = devm_clk_get(&pdev->dev, "apb");
+ if (IS_ERR(scodec->clk_apb)) {
+ dev_err(&pdev->dev, "Failed to get the APB clock\n");
+ return PTR_ERR(scodec->clk_apb);
+ }
+
+ scodec->clk_module = devm_clk_get(&pdev->dev, "codec");
+ if (IS_ERR(scodec->clk_module)) {
+ dev_err(&pdev->dev, "Failed to get the module clock\n");
+ return PTR_ERR(scodec->clk_module);
+ }
+
+ /* Enable the bus clock */
+ if (clk_prepare_enable(scodec->clk_apb)) {
+ dev_err(&pdev->dev, "Failed to enable the APB clock\n");
+ return -EINVAL;
+ }
+
+ /* DMA configuration for TX FIFO */
+ scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA;
+ scodec->playback_dma_data.maxburst = 4;
+ scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+
+ ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec,
+ &sun4i_codec_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our codec\n");
+ goto err_clk_disable;
+ }
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &sun4i_codec_component,
+ &dummy_cpu_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our DAI\n");
+ goto err_unregister_codec;
+ }
+
+ ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register against DMAEngine\n");
+ goto err_unregister_codec;
+ }
+
+ card = sun4i_codec_create_card(&pdev->dev);
+ if (!card) {
+ dev_err(&pdev->dev, "Failed to create our card\n");
+ goto err_unregister_codec;
+ }
+
+ platform_set_drvdata(pdev, card);
+ snd_soc_card_set_drvdata(card, scodec);
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "Failed to register our card\n");
+ goto err_unregister_codec;
+ }
+
+ return 0;
+
+err_unregister_codec:
+ snd_soc_unregister_codec(&pdev->dev);
+err_clk_disable:
+ clk_disable_unprepare(scodec->clk_apb);
+ return ret;
+}
+
+static int sun4i_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+ snd_soc_unregister_card(card);
+ snd_soc_unregister_codec(&pdev->dev);
+ clk_disable_unprepare(scodec->clk_apb);
+
+ return 0;
+}
+
+static struct platform_driver sun4i_codec_driver = {
+ .driver = {
+ .name = "sun4i-codec",
+ .of_match_table = sun4i_codec_of_match,
+ },
+ .probe = sun4i_codec_probe,
+ .remove = sun4i_codec_remove,
+};
+module_platform_driver(sun4i_codec_driver);
+
+MODULE_DESCRIPTION("Allwinner A10 codec driver");
+MODULE_AUTHOR("Emilio López <emilio@elopez.com.ar>");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_AUTHOR("Maxime Ripard <maxime.ripard@free-electrons.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 4e0c0e5..ba9fc09 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -152,6 +152,7 @@
{ .compatible = "stericsson,snd-soc-mop500", },
{},
};
+MODULE_DEVICE_TABLE(of, snd_soc_mop500_match);
static struct platform_driver snd_soc_mop500_driver = {
.driver = {
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index f5df08d..6d5698b 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -522,9 +522,9 @@
slots_active = hweight32(mask);
dev_dbg(dai->dev, "TDM-slots active: %d", slots_active);
- snd_pcm_hw_constraint_minmax(runtime,
+ snd_pcm_hw_constraint_single(runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
- slots_active, slots_active);
+ slots_active);
break;
default:
@@ -843,6 +843,7 @@
{ .compatible = "stericsson,ux500-msp-i2s", },
{},
};
+MODULE_DEVICE_TABLE(of, ux500_msp_i2s_match);
static struct platform_driver msp_i2s_driver = {
.driver = {
diff --git a/sound/usb/card.h b/sound/usb/card.h
index ef580b4..71778ca 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -122,6 +122,7 @@
unsigned int buffer_periods; /* current periods per buffer */
unsigned int altset_idx; /* USB data format: index of alternate setting */
unsigned int txfr_quirk:1; /* allow sub-frame alignment */
+ unsigned int tx_length_quirk:1; /* add length specifier to transfers */
unsigned int fmt_type; /* USB audio format type (1-3) */
unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index e6f7189..7b1cb36 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -183,13 +183,53 @@
ep->retire_data_urb(ep->data_subs, urb);
}
+static void prepare_silent_urb(struct snd_usb_endpoint *ep,
+ struct snd_urb_ctx *ctx)
+{
+ struct urb *urb = ctx->urb;
+ unsigned int offs = 0;
+ unsigned int extra = 0;
+ __le32 packet_length;
+ int i;
+
+ /* For tx_length_quirk, put packet length at start of packet */
+ if (ep->chip->tx_length_quirk)
+ extra = sizeof(packet_length);
+
+ for (i = 0; i < ctx->packets; ++i) {
+ unsigned int offset;
+ unsigned int length;
+ int counts;
+
+ if (ctx->packet_size[i])
+ counts = ctx->packet_size[i];
+ else
+ counts = snd_usb_endpoint_next_packet_size(ep);
+
+ length = counts * ep->stride; /* number of silent bytes */
+ offset = offs * ep->stride + extra * i;
+ urb->iso_frame_desc[i].offset = offset;
+ urb->iso_frame_desc[i].length = length + extra;
+ if (extra) {
+ packet_length = cpu_to_le32(length);
+ memcpy(urb->transfer_buffer + offset,
+ &packet_length, sizeof(packet_length));
+ }
+ memset(urb->transfer_buffer + offset + extra,
+ ep->silence_value, length);
+ offs += counts;
+ }
+
+ urb->number_of_packets = ctx->packets;
+ urb->transfer_buffer_length = offs * ep->stride + ctx->packets * extra;
+}
+
/*
* Prepare a PLAYBACK urb for submission to the bus.
*/
static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
struct snd_urb_ctx *ctx)
{
- int i;
struct urb *urb = ctx->urb;
unsigned char *cp = urb->transfer_buffer;
@@ -201,24 +241,7 @@
ep->prepare_data_urb(ep->data_subs, urb);
} else {
/* no data provider, so send silence */
- unsigned int offs = 0;
- for (i = 0; i < ctx->packets; ++i) {
- int counts;
-
- if (ctx->packet_size[i])
- counts = ctx->packet_size[i];
- else
- counts = snd_usb_endpoint_next_packet_size(ep);
-
- urb->iso_frame_desc[i].offset = offs * ep->stride;
- urb->iso_frame_desc[i].length = counts * ep->stride;
- offs += counts;
- }
-
- urb->number_of_packets = ctx->packets;
- urb->transfer_buffer_length = offs * ep->stride;
- memset(urb->transfer_buffer, ep->silence_value,
- offs * ep->stride);
+ prepare_silent_urb(ep, ctx);
}
break;
@@ -594,6 +617,8 @@
unsigned int max_packs_per_period, urbs_per_period, urb_packs;
unsigned int max_urbs, i;
int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels;
+ int tx_length_quirk = (ep->chip->tx_length_quirk &&
+ usb_pipeout(ep->pipe));
if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) {
/*
@@ -610,13 +635,34 @@
/* assume max. frequency is 25% higher than nominal */
ep->freqmax = ep->freqn + (ep->freqn >> 2);
- maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
- >> (16 - ep->datainterval);
+ /* Round up freqmax to nearest integer in order to calculate maximum
+ * packet size, which must represent a whole number of frames.
+ * This is accomplished by adding 0x0.ffff before converting the
+ * Q16.16 format into integer.
+ * In order to accurately calculate the maximum packet size when
+ * the data interval is more than 1 (i.e. ep->datainterval > 0),
+ * multiply by the data interval prior to rounding. For instance,
+ * a freqmax of 41 kHz will result in a max packet size of 6 (5.125)
+ * frames with a data interval of 1, but 11 (10.25) frames with a
+ * data interval of 2.
+ * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the
+ * maximum datainterval value of 3, at USB full speed, higher for
+ * USB high speed, noting that ep->freqmax is in units of
+ * frames per packet in Q16.16 format.)
+ */
+ maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
+ (frame_bits >> 3);
+ if (tx_length_quirk)
+ maxsize += sizeof(__le32); /* Space for length descriptor */
/* but wMaxPacketSize might reduce this */
if (ep->maxpacksize && ep->maxpacksize < maxsize) {
/* whatever fits into a max. size packet */
- maxsize = ep->maxpacksize;
- ep->freqmax = (maxsize / (frame_bits >> 3))
+ unsigned int data_maxsize = maxsize = ep->maxpacksize;
+
+ if (tx_length_quirk)
+ /* Need to remove the length descriptor to calc freq */
+ data_maxsize -= sizeof(__le32);
+ ep->freqmax = (data_maxsize / (frame_bits >> 3))
<< (16 - ep->datainterval);
}
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 417ebb1..7661616 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1903,11 +1903,14 @@
hostif = &intf->altsetting[1];
intfd = get_iface_desc(hostif);
+ /* If either or both of the endpoints support interrupt transfer,
+ * then use the alternate setting
+ */
if (intfd->bNumEndpoints != 2 ||
- (get_endpoint(hostif, 0)->bmAttributes &
- USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK ||
- (get_endpoint(hostif, 1)->bmAttributes &
- USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ !((get_endpoint(hostif, 0)->bmAttributes &
+ USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT ||
+ (get_endpoint(hostif, 1)->bmAttributes &
+ USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT))
return;
dev_dbg(&umidi->dev->dev, "switching to altsetting %d with int ep\n",
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index d3608c0..fe91184 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -338,7 +338,7 @@
struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
struct usb_mixer_interface *mixer = list->mixer;
int index = kcontrol->private_value & 0xff;
- int value = ucontrol->value.integer.value[0];
+ unsigned int value = ucontrol->value.integer.value[0];
int old_value = kcontrol->private_value >> 8;
int err;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index cdac517..9245f52 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1383,6 +1383,56 @@
subs->hwptr_done++;
}
}
+ if (subs->hwptr_done >= runtime->buffer_size * stride)
+ subs->hwptr_done -= runtime->buffer_size * stride;
+}
+
+static void copy_to_urb(struct snd_usb_substream *subs, struct urb *urb,
+ int offset, int stride, unsigned int bytes)
+{
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+
+ if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
+ /* err, the transferred area goes over buffer boundary. */
+ unsigned int bytes1 =
+ runtime->buffer_size * stride - subs->hwptr_done;
+ memcpy(urb->transfer_buffer + offset,
+ runtime->dma_area + subs->hwptr_done, bytes1);
+ memcpy(urb->transfer_buffer + offset + bytes1,
+ runtime->dma_area, bytes - bytes1);
+ } else {
+ memcpy(urb->transfer_buffer + offset,
+ runtime->dma_area + subs->hwptr_done, bytes);
+ }
+ subs->hwptr_done += bytes;
+ if (subs->hwptr_done >= runtime->buffer_size * stride)
+ subs->hwptr_done -= runtime->buffer_size * stride;
+}
+
+static unsigned int copy_to_urb_quirk(struct snd_usb_substream *subs,
+ struct urb *urb, int stride,
+ unsigned int bytes)
+{
+ __le32 packet_length;
+ int i;
+
+ /* Put __le32 length descriptor at start of each packet. */
+ for (i = 0; i < urb->number_of_packets; i++) {
+ unsigned int length = urb->iso_frame_desc[i].length;
+ unsigned int offset = urb->iso_frame_desc[i].offset;
+
+ packet_length = cpu_to_le32(length);
+ offset += i * sizeof(packet_length);
+ urb->iso_frame_desc[i].offset = offset;
+ urb->iso_frame_desc[i].length += sizeof(packet_length);
+ memcpy(urb->transfer_buffer + offset,
+ &packet_length, sizeof(packet_length));
+ copy_to_urb(subs, urb, offset + sizeof(packet_length),
+ stride, length);
+ }
+ /* Adjust transfer size accordingly. */
+ bytes += urb->number_of_packets * sizeof(packet_length);
+ return bytes;
}
static void prepare_playback_urb(struct snd_usb_substream *subs,
@@ -1460,27 +1510,17 @@
}
subs->hwptr_done += bytes;
+ if (subs->hwptr_done >= runtime->buffer_size * stride)
+ subs->hwptr_done -= runtime->buffer_size * stride;
} else {
/* usual PCM */
- if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
- /* err, the transferred area goes over buffer boundary. */
- unsigned int bytes1 =
- runtime->buffer_size * stride - subs->hwptr_done;
- memcpy(urb->transfer_buffer,
- runtime->dma_area + subs->hwptr_done, bytes1);
- memcpy(urb->transfer_buffer + bytes1,
- runtime->dma_area, bytes - bytes1);
- } else {
- memcpy(urb->transfer_buffer,
- runtime->dma_area + subs->hwptr_done, bytes);
- }
-
- subs->hwptr_done += bytes;
+ if (!subs->tx_length_quirk)
+ copy_to_urb(subs, urb, 0, stride, bytes);
+ else
+ bytes = copy_to_urb_quirk(subs, urb, stride, bytes);
+ /* bytes is now amount of outgoing data */
}
- if (subs->hwptr_done >= runtime->buffer_size * stride)
- subs->hwptr_done -= runtime->buffer_size * stride;
-
/* update delay with exact number of samples queued */
runtime->delay = subs->last_delay;
runtime->delay += frames;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e475665..1a1e2e4 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2664,6 +2664,15 @@
}
},
{
+ USB_DEVICE(0x1235, 0x000a),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "Novation", */
+ /* .product_name = "Nocturn", */
+ .ifnum = 0,
+ .type = QUIRK_MIDI_RAW_BYTES
+ }
+},
+{
USB_DEVICE(0x1235, 0x000e),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
/* .vendor_name = "Novation", */
@@ -3182,10 +3191,9 @@
{
/*
* ZOOM R16/24 in audio interface mode.
- * Mixer descriptors are garbage, further quirks will be needed
- * to make any of it functional, thus disabled for now.
- * Playback stream appears to start and run fine but no sound
- * is produced, so also disabled for now.
+ * Playback requires an extra four byte LE length indicator
+ * at the start of each isochronous packet. This quirk is
+ * enabled in create_standard_audio_quirk().
*/
USB_DEVICE(0x1686, 0x00dd),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
@@ -3193,14 +3201,9 @@
.type = QUIRK_COMPOSITE,
.data = (const struct snd_usb_audio_quirk[]) {
{
- /* Mixer */
- .ifnum = 0,
- .type = QUIRK_IGNORE_INTERFACE,
- },
- {
/* Playback */
.ifnum = 1,
- .type = QUIRK_IGNORE_INTERFACE,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE,
},
{
/* Capture */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 00ebc0c..4897ea1 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -115,6 +115,9 @@
struct usb_interface_descriptor *altsd;
int err;
+ if (chip->usb_id == USB_ID(0x1686, 0x00dd)) /* Zoom R16/24 */
+ chip->tx_length_quirk = 1;
+
alts = &iface->altsetting[0];
altsd = get_iface_desc(alts);
err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber);
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 9700860..8ee14f2 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -92,6 +92,7 @@
subs->direction = stream;
subs->dev = as->chip->dev;
subs->txfr_quirk = as->chip->txfr_quirk;
+ subs->tx_length_quirk = as->chip->tx_length_quirk;
subs->speed = snd_usb_get_speed(subs->dev);
subs->pkt_offset_adj = 0;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 33a1764..15a1271 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -43,6 +43,7 @@
atomic_t usage_count;
wait_queue_head_t shutdown_wait;
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
+ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
int num_interfaces;
int num_suspended_intf;