Hi,
This adds audio support to Motorola Droid 4. I dropped the
regulator from the DT binding as requested by Mark.
Tested:
- Playing via both DACs using Speaker, Earpiece, Headphone
- Recording using internal Mic
- Volume Controls
Known not to work:
- 3.5mm detection support (requires closed source firmware,
needs further investigation)
- Modem / Bluetooth Audio (needs different DAI/I2S config)
Changes since PATCHv3:
* Drop regulator from DT binding, request VAUDIO from
global regulator namespace instead
Changes since PATCHv2:
* Fix a whitespace issue
* Fix const notes Takashi provided
* Fix a DAPM route issue I accidently introduced in v2
Changes since PATCHv1:
* Add patch from Tony exporting soc_dpcm_runtime_update
* Integrate DT patch for vaudio initial mode
* Split dt-binding from codec patch and add Ack from Rob
* Fix CPCAP position in Kconfig/Makefile
* Avoid "err +=" constructs
* Simplify reset function
* Drop cpcap_audio_read/write helpers
* Do not use tertiary operator for mute register value
* Update Input Mux logic
* Switch from simple-audio-card to audio-graph-card
-- Sebastian
Sebastian Reichel (4):
dt-bindings: sound: add motorola,cpcap-audio-codec
ASoC: codec: cpcap: new codec
ARM: dts: motorola-cpcap-mapphone: add audio-codec
ARM: dts: omap4-droid4: add soundcard
.../bindings/sound/motorola,cpcap-audio-codec.txt | 17 +
arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 5 +
arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++
sound/soc/codecs/Kconfig | 5 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/cpcap.c | 1588 ++++++++++++++++++++
6 files changed, 1701 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
create mode 100644 sound/soc/codecs/cpcap.c
--
2.15.1
Add node for audio-codec to its DT file.
Signed-off-by: Sebastian Reichel <[email protected]>
---
arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
index 4d61e5b1334a..69ff53e5702d 100644
--- a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
+++ b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
@@ -68,6 +68,11 @@
};
};
+ cpcap_audio: audio-codec {
+ #sound-dai-cells = <1>;
+ compatible = "motorola,cpcap-audio-codec";
+ };
+
cpcap_rtc: rtc {
compatible = "motorola,cpcap-rtc";
--
2.15.1
Add sound support to Motorola Droid 4 using simple-soundcard
and CPCAP's audio codec. This does not yet correctly represent
the whole audio routing, since McBSP3 is also connected to
Bluetooth and MDM6600 modem (and probably also 4G modem).
These extra DAI links are not yet supported and have not been
tested.
Signed-off-by: Sebastian Reichel <[email protected]>
---
arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++++++++++++++++++++++++++++++++
1 file changed, 84 insertions(+)
diff --git a/arch/arm/boot/dts/omap4-droid4-xt894.dts b/arch/arm/boot/dts/omap4-droid4-xt894.dts
index e11a24397163..bfb0e0ddc269 100644
--- a/arch/arm/boot/dts/omap4-droid4-xt894.dts
+++ b/arch/arm/boot/dts/omap4-droid4-xt894.dts
@@ -118,6 +118,26 @@
};
};
+
+ soundcard {
+ compatible = "audio-graph-card";
+ label = "Droid 4 Audio";
+
+ simple-graph-card,widgets =
+ "Speaker", "Earpiece",
+ "Speaker", "Loudspeaker",
+ "Headphone", "Headphone Jack",
+ "Microphone", "Internal Mic";
+
+ simple-graph-card,routing =
+ "Earpiece", "EP",
+ "Loudspeaker", "SPKR",
+ "Headphone Jack", "HSL",
+ "Headphone Jack", "HSR",
+ "MICR", "Internal Mic";
+
+ dais = <&mcbsp2_port>, <&mcbsp3_port>;
+ };
};
&dss {
@@ -515,6 +535,24 @@
OMAP4_IOPAD(0x112, PIN_OUTPUT_PULLUP | MUX_MODE5) /* uart4_rts */
>;
};
+
+ mcbsp2_pins: pinmux_mcbsp2_pins {
+ pinctrl-single,pins = <
+ OMAP4_IOPAD(0x0f6, PIN_INPUT | MUX_MODE0) /* abe_mcbsp2_clkx */
+ OMAP4_IOPAD(0x0f8, PIN_INPUT | MUX_MODE0) /* abe_mcbsp2_dr */
+ OMAP4_IOPAD(0x0fa, PIN_OUTPUT | MUX_MODE0) /* abe_mcbsp2_dx */
+ OMAP4_IOPAD(0x0fc, PIN_INPUT | MUX_MODE0) /* abe_mcbsp2_fsx */
+ >;
+ };
+
+ mcbsp3_pins: pinmux_mcbsp3_pins {
+ pinctrl-single,pins = <
+ OMAP4_IOPAD(0x106, PIN_INPUT | MUX_MODE1) /* abe_mcbsp3_dr */
+ OMAP4_IOPAD(0x108, PIN_OUTPUT | MUX_MODE1) /* abe_mcbsp3_dx */
+ OMAP4_IOPAD(0x10a, PIN_INPUT | MUX_MODE1) /* abe_mcbsp3_clkx */
+ OMAP4_IOPAD(0x10c, PIN_INPUT | MUX_MODE1) /* abe_mcbsp3_fsx */
+ >;
+ };
};
&omap4_pmx_wkup {
@@ -600,3 +638,49 @@
"0", "0", "1";
};
};
+
+&mcbsp2 {
+ #sound-dai-cells = <0>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&mcbsp2_pins>;
+ status = "okay";
+
+ mcbsp2_port: port {
+ cpu_dai2: endpoint {
+ dai-format = "i2s";
+ remote-endpoint = <&cpcap_audio_codec0>;
+ frame-master = <&cpcap_audio_codec0>;
+ bitclock-master = <&cpcap_audio_codec0>;
+ };
+ };
+};
+
+&mcbsp3 {
+ #sound-dai-cells = <0>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&mcbsp3_pins>;
+ status = "okay";
+
+ mcbsp3_port: port {
+ cpu_dai3: endpoint {
+ dai-format = "dsp_a";
+ frame-master = <&cpcap_audio_codec1>;
+ bitclock-master = <&cpcap_audio_codec1>;
+ remote-endpoint = <&cpcap_audio_codec1>;
+ };
+ };
+};
+
+&cpcap_audio {
+ port@0 {
+ cpcap_audio_codec0: endpoint {
+ remote-endpoint = <&cpu_dai2>;
+ };
+ };
+
+ port@1 {
+ cpcap_audio_codec1: endpoint {
+ remote-endpoint = <&cpu_dai3>;
+ };
+ };
+};
--
2.15.1
Motorola CPCAP is a PMIC with audio functionality, that can be
found on Motorola Droid 4 and probably a few other phones from
Motorola's Droid series.
The driver has been written from scratch using Motorola's Android
driver, register dumps from running Android and datasheet for NXP
MC13783UG (which is similar to Motorola CPCAP, but not the same).
The chip provides two audio interfaces, that can be muxed to two
different audio codecs. One provides support for stereo output
(named StDAC or HiFi), while the other only provides mono output
(named Voice). Only the Voice codec provides a Capture interface.
Signed-off-by: Sebastian Reichel <[email protected]>
---
sound/soc/codecs/Kconfig | 5 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/cpcap.c | 1588 +++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 1595 insertions(+)
create mode 100644 sound/soc/codecs/cpcap.c
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 2b331f7266ab..25be0c64bb1a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -408,6 +408,11 @@ config SND_SOC_ALC5632
config SND_SOC_BT_SCO
tristate "Dummy BT SCO codec driver"
+config SND_SOC_CPCAP
+ tristate "Motorola CPCAP codec"
+ depends on MFD_CPCAP
+ default MFD_CPCAP
+
config SND_SOC_CQ0093VC
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index da1571336f1e..2aeee1ba034e 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -36,6 +36,7 @@ snd-soc-ak4671-objs := ak4671.o
snd-soc-ak5386-objs := ak5386.o
snd-soc-arizona-objs := arizona.o
snd-soc-bt-sco-objs := bt-sco.o
+snd-soc-cpcap-objs := cpcap.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs35l32-objs := cs35l32.o
snd-soc-cs35l33-objs := cs35l33.o
@@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o
obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o
obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
+obj-$(CONFIG_SND_SOC_CPCAP) += snd-soc-cpcap.o
obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o
obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o
obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
new file mode 100644
index 000000000000..0d5a6ce7978f
--- /dev/null
+++ b/sound/soc/codecs/cpcap.c
@@ -0,0 +1,1588 @@
+/*
+ * ALSA SoC CPCAP codec driver
+ *
+ * Copyright (C) 2017 - 2018 Sebastian Reichel <[email protected]>
+ *
+ * Very loosely based on original driver from Motorola:
+ * Copyright (C) 2007 - 2009 Motorola, Inc.
+ *
+ * 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/regmap.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/motorola-cpcap.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+/* Register 513 CPCAP_REG_CC --- CODEC */
+#define CPCAP_BIT_CDC_CLK2 15
+#define CPCAP_BIT_CDC_CLK1 14
+#define CPCAP_BIT_CDC_CLK0 13
+#define CPCAP_BIT_CDC_SR3 12
+#define CPCAP_BIT_CDC_SR2 11
+#define CPCAP_BIT_CDC_SR1 10
+#define CPCAP_BIT_CDC_SR0 9
+#define CPCAP_BIT_CDC_CLOCK_TREE_RESET 8
+#define CPCAP_BIT_MIC2_CDC_EN 7
+#define CPCAP_BIT_CDC_EN_RX 6
+#define CPCAP_BIT_DF_RESET 5
+#define CPCAP_BIT_MIC1_CDC_EN 4
+#define CPCAP_BIT_AUDOHPF_1 3
+#define CPCAP_BIT_AUDOHPF_0 2
+#define CPCAP_BIT_AUDIHPF_1 1
+#define CPCAP_BIT_AUDIHPF_0 0
+
+/* Register 514 CPCAP_REG_CDI --- CODEC Digital Audio Interface */
+#define CPCAP_BIT_CDC_PLL_SEL 15
+#define CPCAP_BIT_CLK_IN_SEL 13
+#define CPCAP_BIT_DIG_AUD_IN 12
+#define CPCAP_BIT_CDC_CLK_EN 11
+#define CPCAP_BIT_CDC_DIG_AUD_FS1 10
+#define CPCAP_BIT_CDC_DIG_AUD_FS0 9
+#define CPCAP_BIT_MIC2_TIMESLOT2 8
+#define CPCAP_BIT_MIC2_TIMESLOT1 7
+#define CPCAP_BIT_MIC2_TIMESLOT0 6
+#define CPCAP_BIT_MIC1_RX_TIMESLOT2 5
+#define CPCAP_BIT_MIC1_RX_TIMESLOT1 4
+#define CPCAP_BIT_MIC1_RX_TIMESLOT0 3
+#define CPCAP_BIT_FS_INV 2
+#define CPCAP_BIT_CLK_INV 1
+#define CPCAP_BIT_SMB_CDC 0
+
+/* Register 515 CPCAP_REG_SDAC --- Stereo DAC */
+#define CPCAP_BIT_FSYNC_CLK_IN_COMMON 11
+#define CPCAP_BIT_SLAVE_PLL_CLK_INPUT 10
+#define CPCAP_BIT_ST_CLOCK_TREE_RESET 9
+#define CPCAP_BIT_DF_RESET_ST_DAC 8
+#define CPCAP_BIT_ST_SR3 7
+#define CPCAP_BIT_ST_SR2 6
+#define CPCAP_BIT_ST_SR1 5
+#define CPCAP_BIT_ST_SR0 4
+#define CPCAP_BIT_ST_DAC_CLK2 3
+#define CPCAP_BIT_ST_DAC_CLK1 2
+#define CPCAP_BIT_ST_DAC_CLK0 1
+#define CPCAP_BIT_ST_DAC_EN 0
+
+/* Register 516 CPCAP_REG_SDACDI --- Stereo DAC Digital Audio Interface */
+#define CPCAP_BIT_ST_L_TIMESLOT2 13
+#define CPCAP_BIT_ST_L_TIMESLOT1 12
+#define CPCAP_BIT_ST_L_TIMESLOT0 11
+#define CPCAP_BIT_ST_R_TIMESLOT2 10
+#define CPCAP_BIT_ST_R_TIMESLOT1 9
+#define CPCAP_BIT_ST_R_TIMESLOT0 8
+#define CPCAP_BIT_ST_DAC_CLK_IN_SEL 7
+#define CPCAP_BIT_ST_FS_INV 6
+#define CPCAP_BIT_ST_CLK_INV 5
+#define CPCAP_BIT_ST_DIG_AUD_FS1 4
+#define CPCAP_BIT_ST_DIG_AUD_FS0 3
+#define CPCAP_BIT_DIG_AUD_IN_ST_DAC 2
+#define CPCAP_BIT_ST_CLK_EN 1
+#define CPCAP_BIT_SMB_ST_DAC 0
+
+/* Register 517 CPCAP_REG_TXI --- TX Interface */
+#define CPCAP_BIT_PTT_TH 15
+#define CPCAP_BIT_PTT_CMP_EN 14
+#define CPCAP_BIT_HS_ID_TX 13
+#define CPCAP_BIT_MB_ON2 12
+#define CPCAP_BIT_MB_ON1L 11
+#define CPCAP_BIT_MB_ON1R 10
+#define CPCAP_BIT_RX_L_ENCODE 9
+#define CPCAP_BIT_RX_R_ENCODE 8
+#define CPCAP_BIT_MIC2_MUX 7
+#define CPCAP_BIT_MIC2_PGA_EN 6
+#define CPCAP_BIT_CDET_DIS 5
+#define CPCAP_BIT_EMU_MIC_MUX 4
+#define CPCAP_BIT_HS_MIC_MUX 3
+#define CPCAP_BIT_MIC1_MUX 2
+#define CPCAP_BIT_MIC1_PGA_EN 1
+#define CPCAP_BIT_DLM 0
+
+/* Register 518 CPCAP_REG_TXMP --- Mic Gain */
+#define CPCAP_BIT_MB_BIAS_R1 11
+#define CPCAP_BIT_MB_BIAS_R0 10
+#define CPCAP_BIT_MIC2_GAIN_4 9
+#define CPCAP_BIT_MIC2_GAIN_3 8
+#define CPCAP_BIT_MIC2_GAIN_2 7
+#define CPCAP_BIT_MIC2_GAIN_1 6
+#define CPCAP_BIT_MIC2_GAIN_0 5
+#define CPCAP_BIT_MIC1_GAIN_4 4
+#define CPCAP_BIT_MIC1_GAIN_3 3
+#define CPCAP_BIT_MIC1_GAIN_2 2
+#define CPCAP_BIT_MIC1_GAIN_1 1
+#define CPCAP_BIT_MIC1_GAIN_0 0
+
+/* Register 519 CPCAP_REG_RXOA --- RX Output Amplifier */
+#define CPCAP_BIT_UNUSED_519_15 15
+#define CPCAP_BIT_UNUSED_519_14 14
+#define CPCAP_BIT_UNUSED_519_13 13
+#define CPCAP_BIT_STDAC_LOW_PWR_DISABLE 12
+#define CPCAP_BIT_HS_LOW_PWR 11
+#define CPCAP_BIT_HS_ID_RX 10
+#define CPCAP_BIT_ST_HS_CP_EN 9
+#define CPCAP_BIT_EMU_SPKR_R_EN 8
+#define CPCAP_BIT_EMU_SPKR_L_EN 7
+#define CPCAP_BIT_HS_L_EN 6
+#define CPCAP_BIT_HS_R_EN 5
+#define CPCAP_BIT_A4_LINEOUT_L_EN 4
+#define CPCAP_BIT_A4_LINEOUT_R_EN 3
+#define CPCAP_BIT_A2_LDSP_L_EN 2
+#define CPCAP_BIT_A2_LDSP_R_EN 1
+#define CPCAP_BIT_A1_EAR_EN 0
+
+/* Register 520 CPCAP_REG_RXVC --- RX Volume Control */
+#define CPCAP_BIT_VOL_EXT3 15
+#define CPCAP_BIT_VOL_EXT2 14
+#define CPCAP_BIT_VOL_EXT1 13
+#define CPCAP_BIT_VOL_EXT0 12
+#define CPCAP_BIT_VOL_DAC3 11
+#define CPCAP_BIT_VOL_DAC2 10
+#define CPCAP_BIT_VOL_DAC1 9
+#define CPCAP_BIT_VOL_DAC0 8
+#define CPCAP_BIT_VOL_DAC_LSB_1dB1 7
+#define CPCAP_BIT_VOL_DAC_LSB_1dB0 6
+#define CPCAP_BIT_VOL_CDC3 5
+#define CPCAP_BIT_VOL_CDC2 4
+#define CPCAP_BIT_VOL_CDC1 3
+#define CPCAP_BIT_VOL_CDC0 2
+#define CPCAP_BIT_VOL_CDC_LSB_1dB1 1
+#define CPCAP_BIT_VOL_CDC_LSB_1dB0 0
+
+/* Register 521 CPCAP_REG_RXCOA --- Codec to Output Amp Switches */
+#define CPCAP_BIT_PGA_CDC_EN 10
+#define CPCAP_BIT_CDC_SW 9
+#define CPCAP_BIT_PGA_OUTR_USBDP_CDC_SW 8
+#define CPCAP_BIT_PGA_OUTL_USBDN_CDC_SW 7
+#define CPCAP_BIT_ALEFT_HS_CDC_SW 6
+#define CPCAP_BIT_ARIGHT_HS_CDC_SW 5
+#define CPCAP_BIT_A4_LINEOUT_L_CDC_SW 4
+#define CPCAP_BIT_A4_LINEOUT_R_CDC_SW 3
+#define CPCAP_BIT_A2_LDSP_L_CDC_SW 2
+#define CPCAP_BIT_A2_LDSP_R_CDC_SW 1
+#define CPCAP_BIT_A1_EAR_CDC_SW 0
+
+/* Register 522 CPCAP_REG_RXSDOA --- RX Stereo DAC to Output Amp Switches */
+#define CPCAP_BIT_PGA_DAC_EN 12
+#define CPCAP_BIT_ST_DAC_SW 11
+#define CPCAP_BIT_MONO_DAC1 10
+#define CPCAP_BIT_MONO_DAC0 9
+#define CPCAP_BIT_PGA_OUTR_USBDP_DAC_SW 8
+#define CPCAP_BIT_PGA_OUTL_USBDN_DAC_SW 7
+#define CPCAP_BIT_ALEFT_HS_DAC_SW 6
+#define CPCAP_BIT_ARIGHT_HS_DAC_SW 5
+#define CPCAP_BIT_A4_LINEOUT_L_DAC_SW 4
+#define CPCAP_BIT_A4_LINEOUT_R_DAC_SW 3
+#define CPCAP_BIT_A2_LDSP_L_DAC_SW 2
+#define CPCAP_BIT_A2_LDSP_R_DAC_SW 1
+#define CPCAP_BIT_A1_EAR_DAC_SW 0
+
+/* Register 523 CPCAP_REG_RXEPOA --- RX External PGA to Output Amp Switches */
+#define CPCAP_BIT_PGA_EXT_L_EN 14
+#define CPCAP_BIT_PGA_EXT_R_EN 13
+#define CPCAP_BIT_PGA_IN_L_SW 12
+#define CPCAP_BIT_PGA_IN_R_SW 11
+#define CPCAP_BIT_MONO_EXT1 10
+#define CPCAP_BIT_MONO_EXT0 9
+#define CPCAP_BIT_PGA_OUTR_USBDP_EXT_SW 8
+#define CPCAP_BIT_PGA_OUTL_USBDN_EXT_SW 7
+#define CPCAP_BIT_ALEFT_HS_EXT_SW 6
+#define CPCAP_BIT_ARIGHT_HS_EXT_SW 5
+#define CPCAP_BIT_A4_LINEOUT_L_EXT_SW 4
+#define CPCAP_BIT_A4_LINEOUT_R_EXT_SW 3
+#define CPCAP_BIT_A2_LDSP_L_EXT_SW 2
+#define CPCAP_BIT_A2_LDSP_R_EXT_SW 1
+#define CPCAP_BIT_A1_EAR_EXT_SW 0
+
+/* Register 525 CPCAP_REG_A2LA --- SPK Amplifier and Clock Config for Headset */
+#define CPCAP_BIT_NCP_CLK_SYNC 7
+#define CPCAP_BIT_A2_CLK_SYNC 6
+#define CPCAP_BIT_A2_FREE_RUN 5
+#define CPCAP_BIT_A2_CLK2 4
+#define CPCAP_BIT_A2_CLK1 3
+#define CPCAP_BIT_A2_CLK0 2
+#define CPCAP_BIT_A2_CLK_IN 1
+#define CPCAP_BIT_A2_CONFIG 0
+
+#define SLEEP_ACTIVATE_POWER 2
+#define CLOCK_TREE_RESET_TIME 1
+
+/* constants for ST delay workaround */
+#define STM_STDAC_ACTIVATE_RAMP_TIME 1
+#define STM_STDAC_EN_TEST_PRE 0x090C
+#define STM_STDAC_EN_TEST_POST 0x0000
+#define STM_STDAC_EN_ST_TEST1_PRE 0x2400
+#define STM_STDAC_EN_ST_TEST1_POST 0x0400
+
+struct cpcap_reg_info {
+ u16 reg;
+ u16 mask;
+ u16 val;
+};
+
+static const struct cpcap_reg_info cpcap_default_regs[] = {
+ { CPCAP_REG_CC, 0xFFFF, 0x0000 },
+ { CPCAP_REG_CC, 0xFFFF, 0x0000 },
+ { CPCAP_REG_CDI, 0xBFFF, 0x0000 },
+ { CPCAP_REG_SDAC, 0x0FFF, 0x0000 },
+ { CPCAP_REG_SDACDI, 0x3FFF, 0x0000 },
+ { CPCAP_REG_TXI, 0x0FDF, 0x0000 },
+ { CPCAP_REG_TXMP, 0x0FFF, 0x0400 },
+ { CPCAP_REG_RXOA, 0x01FF, 0x0000 },
+ { CPCAP_REG_RXVC, 0xFF3C, 0x0000 },
+ { CPCAP_REG_RXCOA, 0x07FF, 0x0000 },
+ { CPCAP_REG_RXSDOA, 0x1FFF, 0x0000 },
+ { CPCAP_REG_RXEPOA, 0x7FFF, 0x0000 },
+ { CPCAP_REG_A2LA, BIT(CPCAP_BIT_A2_FREE_RUN),
+ BIT(CPCAP_BIT_A2_FREE_RUN) },
+};
+
+enum cpcap_dai {
+ CPCAP_DAI_HIFI,
+ CPCAP_DAI_VOICE,
+};
+
+struct cpcap_audio {
+ struct snd_soc_codec *codec;
+ struct regmap *regmap;
+
+ u16 vendor;
+
+ int codec_clk_id;
+ int codec_freq;
+ int codec_format;
+};
+
+static int cpcap_st_workaround(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 cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int err = 0;
+
+ /* Only CPCAP from ST requires workaround */
+ if (cpcap->vendor != CPCAP_VENDOR_ST)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ err = regmap_write(cpcap->regmap, CPCAP_REG_TEST,
+ STM_STDAC_EN_TEST_PRE);
+ if (err)
+ return err;
+ err = regmap_write(cpcap->regmap, CPCAP_REG_ST_TEST1,
+ STM_STDAC_EN_ST_TEST1_PRE);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ msleep(STM_STDAC_ACTIVATE_RAMP_TIME);
+
+ err = regmap_write(cpcap->regmap, CPCAP_REG_ST_TEST1,
+ STM_STDAC_EN_ST_TEST1_POST);
+ if (err)
+ return err;
+ err = regmap_write(cpcap->regmap, CPCAP_REG_TEST,
+ STM_STDAC_EN_TEST_POST);
+ break;
+ default:
+ break;
+ }
+
+ return err;
+}
+
+/* Capture Gain Control: 0dB to 31dB in 1dB steps */
+static const DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0);
+
+/* Playback Gain Control: -33dB to 12dB in 3dB steps */
+static const DECLARE_TLV_DB_SCALE(vol_tlv, -3300, 300, 0);
+
+static const char * const cpcap_phase_invert_texts[] = {
+ "Non Inverted", "Inverted"
+};
+static SOC_ENUM_SINGLE_DECL(cpcap_hifi_l_phase_inv_enum,
+ CPCAP_REG_RXSDOA, CPCAP_BIT_MONO_DAC0, cpcap_phase_invert_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_ext_l_phase_inv_enum,
+ CPCAP_REG_RXEPOA, CPCAP_BIT_MONO_EXT0, cpcap_phase_invert_texts);
+
+static const struct snd_kcontrol_new cpcap_snd_controls[] = {
+ /* Playback Gain */
+ SOC_SINGLE_TLV("HiFi Playback Volume",
+ CPCAP_REG_RXVC, CPCAP_BIT_VOL_DAC0, 0xF, 0, vol_tlv),
+ SOC_SINGLE_TLV("Voice Playback Volume",
+ CPCAP_REG_RXVC, CPCAP_BIT_VOL_CDC0, 0xF, 0, vol_tlv),
+ SOC_SINGLE_TLV("Ext Playback Volume",
+ CPCAP_REG_RXVC, CPCAP_BIT_VOL_EXT0, 0xF, 0, vol_tlv),
+
+ /* Capture Gain */
+ SOC_SINGLE_TLV("Mic1 Capture Volume",
+ CPCAP_REG_TXMP, CPCAP_BIT_MIC1_GAIN_0, 0x1F, 0, mic_gain_tlv),
+ SOC_SINGLE_TLV("Mic2 Capture Volume",
+ CPCAP_REG_TXMP, CPCAP_BIT_MIC2_GAIN_0, 0x1F, 0, mic_gain_tlv),
+
+ SOC_ENUM("Hifi Left Phase Playback Switch",
+ cpcap_hifi_l_phase_inv_enum),
+ SOC_ENUM("Ext Left Phase Playback Switch",
+ cpcap_ext_l_phase_inv_enum),
+};
+
+static const char * const cpcap_out_mux_texts[] = {
+ "Off", "Voice", "HiFi", "Ext"
+};
+
+static const char * const cpcap_in_right_mux_texts[] = {
+ "Off", "Mic 1", "Headset Mic", "EMU Mic", "Ext Right"
+};
+
+static const char * const cpcap_in_left_mux_texts[] = {
+ "Off", "Mic 2", "Ext Left"
+};
+
+/*
+ * input muxes use unusual register layout, so that we need to use custom
+ * getter/setter methods
+ */
+static SOC_ENUM_SINGLE_EXT_DECL(cpcap_input_left_mux_enum,
+ cpcap_in_left_mux_texts);
+static SOC_ENUM_SINGLE_EXT_DECL(cpcap_input_right_mux_enum,
+ cpcap_in_right_mux_texts);
+
+/*
+ * mux uses same bit in CPCAP_REG_RXCOA, CPCAP_REG_RXSDOA & CPCAP_REG_RXEPOA;
+ * even though the register layout makes it look like a mixer, this is a mux.
+ * Enabling multiple inputs will result in no audio being forwarded.
+ */
+static SOC_ENUM_SINGLE_DECL(cpcap_earpiece_mux_enum, 0, 0, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_spkr_r_mux_enum, 0, 1, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_spkr_l_mux_enum, 0, 2, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_line_r_mux_enum, 0, 3, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_line_l_mux_enum, 0, 4, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_hs_r_mux_enum, 0, 5, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_hs_l_mux_enum, 0, 6, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_emu_l_mux_enum, 0, 7, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_emu_r_mux_enum, 0, 8, cpcap_out_mux_texts);
+
+static int cpcap_output_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int shift = e->shift_l;
+ int reg_voice, reg_hifi, reg_ext, status;
+ int err;
+
+ err = regmap_read(cpcap->regmap, CPCAP_REG_RXCOA, ®_voice);
+ if (err)
+ return err;
+ err = regmap_read(cpcap->regmap, CPCAP_REG_RXSDOA, ®_hifi);
+ if (err)
+ return err;
+ err = regmap_read(cpcap->regmap, CPCAP_REG_RXEPOA, ®_ext);
+ if (err)
+ return err;
+
+ reg_voice = (reg_voice >> shift) & 1;
+ reg_hifi = (reg_hifi >> shift) & 1;
+ reg_ext = (reg_ext >> shift) & 1;
+ status = reg_ext << 2 | reg_hifi << 1 | reg_voice;
+
+ switch (status) {
+ case 0x04:
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case 0x02:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case 0x01:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ }
+
+ return 0;
+}
+
+static int cpcap_output_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int muxval = ucontrol->value.enumerated.item[0];
+ unsigned int mask = BIT(e->shift_l);
+ u16 reg_voice = 0x00, reg_hifi = 0x00, reg_ext = 0x00;
+ int err;
+
+ switch (muxval) {
+ case 1:
+ reg_voice = mask;
+ break;
+ case 2:
+ reg_hifi = mask;
+ break;
+ case 3:
+ reg_ext = mask;
+ break;
+ default:
+ break;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA,
+ mask, reg_voice);
+ if (err)
+ return err;
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXSDOA,
+ mask, reg_hifi);
+ if (err)
+ return err;
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXEPOA,
+ mask, reg_ext);
+ if (err)
+ return err;
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, muxval, e, NULL);
+
+ return 0;
+}
+
+static int cpcap_input_right_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int regval, mask;
+ int err;
+
+ err = regmap_read(cpcap->regmap, CPCAP_REG_TXI, ®val);
+ if (err)
+ return err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC1_MUX);
+ mask |= BIT(CPCAP_BIT_HS_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_EMU_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_RX_R_ENCODE);
+
+ switch (regval & mask) {
+ case BIT(CPCAP_BIT_RX_R_ENCODE):
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ case BIT(CPCAP_BIT_EMU_MIC_MUX):
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case BIT(CPCAP_BIT_HS_MIC_MUX):
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case BIT(CPCAP_BIT_MIC1_MUX):
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ }
+
+ return 0;
+}
+
+static int cpcap_input_right_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int muxval = ucontrol->value.enumerated.item[0];
+ int regval = 0, mask;
+ int err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC1_MUX);
+ mask |= BIT(CPCAP_BIT_HS_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_EMU_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_RX_R_ENCODE);
+
+ switch (muxval) {
+ case 1:
+ regval = BIT(CPCAP_BIT_MIC1_MUX);
+ break;
+ case 2:
+ regval = BIT(CPCAP_BIT_HS_MIC_MUX);
+ break;
+ case 3:
+ regval = BIT(CPCAP_BIT_EMU_MIC_MUX);
+ break;
+ case 4:
+ regval = BIT(CPCAP_BIT_RX_R_ENCODE);
+ break;
+ default:
+ break;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+ mask, regval);
+ if (err)
+ return err;
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, muxval, e, NULL);
+
+ return 0;
+}
+
+static int cpcap_input_left_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int regval, mask;
+ int err;
+
+ err = regmap_read(cpcap->regmap, CPCAP_REG_TXI, ®val);
+ if (err)
+ return err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC2_MUX);
+ mask |= BIT(CPCAP_BIT_RX_L_ENCODE);
+
+ switch (regval & mask) {
+ case BIT(CPCAP_BIT_RX_L_ENCODE):
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case BIT(CPCAP_BIT_MIC2_MUX):
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ }
+
+ return 0;
+}
+
+static int cpcap_input_left_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int muxval = ucontrol->value.enumerated.item[0];
+ int regval = 0, mask;
+ int err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC2_MUX);
+ mask |= BIT(CPCAP_BIT_RX_L_ENCODE);
+
+ switch (muxval) {
+ case 1:
+ regval = BIT(CPCAP_BIT_MIC2_MUX);
+ break;
+ case 2:
+ regval = BIT(CPCAP_BIT_RX_L_ENCODE);
+ break;
+ default:
+ break;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+ mask, regval);
+ if (err)
+ return err;
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, muxval, e, NULL);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new cpcap_input_left_mux =
+ SOC_DAPM_ENUM_EXT("Input Left", cpcap_input_left_mux_enum,
+ cpcap_input_left_mux_get_enum,
+ cpcap_input_left_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_input_right_mux =
+ SOC_DAPM_ENUM_EXT("Input Right", cpcap_input_right_mux_enum,
+ cpcap_input_right_mux_get_enum,
+ cpcap_input_right_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_emu_left_mux =
+ SOC_DAPM_ENUM_EXT("EMU Left", cpcap_emu_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_emu_right_mux =
+ SOC_DAPM_ENUM_EXT("EMU Right", cpcap_emu_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_hs_left_mux =
+ SOC_DAPM_ENUM_EXT("Headset Left", cpcap_hs_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_hs_right_mux =
+ SOC_DAPM_ENUM_EXT("Headset Right", cpcap_hs_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_line_left_mux =
+ SOC_DAPM_ENUM_EXT("Line Left", cpcap_line_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_line_right_mux =
+ SOC_DAPM_ENUM_EXT("Line Right", cpcap_line_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_speaker_left_mux =
+ SOC_DAPM_ENUM_EXT("Speaker Left", cpcap_spkr_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_speaker_right_mux =
+ SOC_DAPM_ENUM_EXT("Speaker Right", cpcap_spkr_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_earpiece_mux =
+ SOC_DAPM_ENUM_EXT("Earpiece", cpcap_earpiece_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+
+static const struct snd_kcontrol_new cpcap_hifi_mono_mixer_controls[] = {
+ SOC_DAPM_SINGLE("HiFi Mono Playback Switch",
+ CPCAP_REG_RXSDOA, CPCAP_BIT_MONO_DAC1, 1, 0),
+};
+static const struct snd_kcontrol_new cpcap_ext_mono_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Ext Mono Playback Switch",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_MONO_EXT0, 1, 0),
+};
+
+static const struct snd_kcontrol_new cpcap_extr_mute_control =
+ SOC_DAPM_SINGLE("Switch",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_IN_R_SW, 1, 0);
+static const struct snd_kcontrol_new cpcap_extl_mute_control =
+ SOC_DAPM_SINGLE("Switch",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_IN_L_SW, 1, 0);
+
+static const struct snd_kcontrol_new cpcap_voice_loopback =
+ SOC_DAPM_SINGLE("Switch",
+ CPCAP_REG_TXI, CPCAP_BIT_DLM, 1, 0);
+
+static const struct snd_soc_dapm_widget cpcap_dapm_widgets[] = {
+ /* DAIs */
+ SND_SOC_DAPM_AIF_IN("HiFi RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("Voice RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("Voice TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+ /* Power Supply */
+ SND_SOC_DAPM_REGULATOR_SUPPLY("VAUDIO", SLEEP_ACTIVATE_POWER, 0),
+
+ /* Highpass Filters */
+ SND_SOC_DAPM_REG(snd_soc_dapm_pga, "Highpass Filter RX",
+ CPCAP_REG_CC, CPCAP_BIT_AUDIHPF_0, 0x3, 0x3, 0x0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_pga, "Highpass Filter TX",
+ CPCAP_REG_CC, CPCAP_BIT_AUDOHPF_0, 0x3, 0x3, 0x0),
+
+ /* Clocks */
+ SND_SOC_DAPM_SUPPLY("HiFi DAI Clock",
+ CPCAP_REG_SDACDI, CPCAP_BIT_ST_CLK_EN, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Voice DAI Clock",
+ CPCAP_REG_CDI, CPCAP_BIT_CDC_CLK_EN, 0, NULL, 0),
+
+ /* Microphone Bias */
+ SND_SOC_DAPM_SUPPLY("MIC1R Bias",
+ CPCAP_REG_TXI, CPCAP_BIT_MB_ON1R, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC1L Bias",
+ CPCAP_REG_TXI, CPCAP_BIT_MB_ON1L, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC2 Bias",
+ CPCAP_REG_TXI, CPCAP_BIT_MB_ON2, 0, NULL, 0),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MICR"),
+ SND_SOC_DAPM_INPUT("HSMIC"),
+ SND_SOC_DAPM_INPUT("EMUMIC"),
+ SND_SOC_DAPM_INPUT("MICL"),
+ SND_SOC_DAPM_INPUT("EXTR"),
+ SND_SOC_DAPM_INPUT("EXTL"),
+
+ /* Capture Route */
+ SND_SOC_DAPM_MUX("Right Capture Route",
+ SND_SOC_NOPM, 0, 0, &cpcap_input_right_mux),
+ SND_SOC_DAPM_MUX("Left Capture Route",
+ SND_SOC_NOPM, 0, 0, &cpcap_input_left_mux),
+
+ /* Capture PGAs */
+ SND_SOC_DAPM_PGA("Microphone 1 PGA",
+ CPCAP_REG_TXI, CPCAP_BIT_MIC1_PGA_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Microphone 2 PGA",
+ CPCAP_REG_TXI, CPCAP_BIT_MIC2_PGA_EN, 0, NULL, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC Right", NULL,
+ CPCAP_REG_CC, CPCAP_BIT_MIC1_CDC_EN, 0),
+ SND_SOC_DAPM_ADC("ADC Left", NULL,
+ CPCAP_REG_CC, CPCAP_BIT_MIC2_CDC_EN, 0),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC_E("DAC HiFi", NULL,
+ CPCAP_REG_SDAC, CPCAP_BIT_ST_DAC_EN, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_DAC_E("DAC Voice", NULL,
+ CPCAP_REG_CC, CPCAP_BIT_CDC_EN_RX, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+ /* Playback PGA */
+ SND_SOC_DAPM_PGA("HiFi PGA",
+ CPCAP_REG_RXSDOA, CPCAP_BIT_PGA_DAC_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Voice PGA",
+ CPCAP_REG_RXCOA, CPCAP_BIT_PGA_CDC_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("Ext Right PGA",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_EXT_R_EN, 0,
+ NULL, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_E("Ext Left PGA",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_EXT_L_EN, 0,
+ NULL, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+ /* Playback Switch */
+ SND_SOC_DAPM_SWITCH("Ext Right Enable", SND_SOC_NOPM, 0, 0,
+ &cpcap_extr_mute_control),
+ SND_SOC_DAPM_SWITCH("Ext Left Enable", SND_SOC_NOPM, 0, 0,
+ &cpcap_extl_mute_control),
+
+ /* Loopback Switch */
+ SND_SOC_DAPM_SWITCH("Voice Loopback", SND_SOC_NOPM, 0, 0,
+ &cpcap_voice_loopback),
+
+ /* Mono Mixer */
+ SOC_MIXER_ARRAY("HiFi Mono Left Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_hifi_mono_mixer_controls),
+ SOC_MIXER_ARRAY("HiFi Mono Right Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_hifi_mono_mixer_controls),
+ SOC_MIXER_ARRAY("Ext Mono Left Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_ext_mono_mixer_controls),
+ SOC_MIXER_ARRAY("Ext Mono Right Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_ext_mono_mixer_controls),
+
+ /* Output Routes */
+ SND_SOC_DAPM_MUX("Earpiece Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_earpiece_mux),
+ SND_SOC_DAPM_MUX("Speaker Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_speaker_right_mux),
+ SND_SOC_DAPM_MUX("Speaker Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_speaker_left_mux),
+ SND_SOC_DAPM_MUX("Lineout Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_line_right_mux),
+ SND_SOC_DAPM_MUX("Lineout Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_line_left_mux),
+ SND_SOC_DAPM_MUX("Headset Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_hs_right_mux),
+ SND_SOC_DAPM_MUX("Headset Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_hs_left_mux),
+ SND_SOC_DAPM_MUX("EMU Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_emu_right_mux),
+ SND_SOC_DAPM_MUX("EMU Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_emu_left_mux),
+
+ /* Output Amplifier */
+ SND_SOC_DAPM_PGA("Earpiece PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A1_EAR_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Speaker Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A2_LDSP_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Speaker Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A2_LDSP_L_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Lineout Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A4_LINEOUT_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Lineout Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A4_LINEOUT_L_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Headset Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_HS_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Headset Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_HS_L_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("EMU Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_EMU_SPKR_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("EMU Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_EMU_SPKR_L_EN, 0, NULL, 0),
+
+ /* Headet Charge Pump */
+ SND_SOC_DAPM_SUPPLY("Headset Charge Pump",
+ CPCAP_REG_RXOA, CPCAP_BIT_ST_HS_CP_EN, 0, NULL, 0),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("EP"),
+ SND_SOC_DAPM_OUTPUT("SPKR"),
+ SND_SOC_DAPM_OUTPUT("SPKL"),
+ SND_SOC_DAPM_OUTPUT("LINER"),
+ SND_SOC_DAPM_OUTPUT("LINEL"),
+ SND_SOC_DAPM_OUTPUT("HSR"),
+ SND_SOC_DAPM_OUTPUT("HSL"),
+ SND_SOC_DAPM_OUTPUT("EMUR"),
+ SND_SOC_DAPM_OUTPUT("EMUL"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* Power Supply */
+ {"HiFi PGA", NULL, "VAUDIO"},
+ {"Voice PGA", NULL, "VAUDIO"},
+ {"Ext Right PGA", NULL, "VAUDIO"},
+ {"Ext Left PGA", NULL, "VAUDIO"},
+ {"Microphone 1 PGA", NULL, "VAUDIO"},
+ {"Microphone 2 PGA", NULL, "VAUDIO"},
+
+ /* Stream -> AIF */
+ {"HiFi RX", NULL, "HiFi Playback"},
+ {"Voice RX", NULL, "Voice Playback"},
+ {"Voice Capture", NULL, "Voice TX"},
+
+ /* AIF clocks */
+ {"HiFi RX", NULL, "HiFi DAI Clock"},
+ {"Voice RX", NULL, "Voice DAI Clock"},
+ {"Voice TX", NULL, "Voice DAI Clock"},
+
+ /* Digital Loopback */
+ {"Voice Loopback", "Switch", "Voice TX"},
+ {"Voice RX", NULL, "Voice Loopback"},
+
+ /* Highpass Filters */
+ {"Highpass Filter RX", NULL, "Voice RX"},
+ {"Voice TX", NULL, "Highpass Filter TX"},
+
+ /* AIF -> DAC mapping */
+ {"DAC HiFi", NULL, "HiFi RX"},
+ {"DAC Voice", NULL, "Highpass Filter RX"},
+
+ /* DAC -> PGA */
+ {"HiFi PGA", NULL, "DAC HiFi"},
+ {"Voice PGA", NULL, "DAC Voice"},
+
+ /* Ext Input -> PGA */
+ {"Ext Right PGA", NULL, "EXTR"},
+ {"Ext Left PGA", NULL, "EXTL"},
+
+ /* Ext PGA -> Ext Playback Switch */
+ {"Ext Right Enable", "Switch", "Ext Right PGA"},
+ {"Ext Left Enable", "Switch", "Ext Left PGA"},
+
+ /* HiFi PGA -> Mono Mixer */
+ {"HiFi Mono Left Mixer", NULL, "HiFi PGA"},
+ {"HiFi Mono Left Mixer", "HiFi Mono Playback Switch", "HiFi PGA"},
+ {"HiFi Mono Right Mixer", NULL, "HiFi PGA"},
+ {"HiFi Mono Right Mixer", "HiFi Mono Playback Switch", "HiFi PGA"},
+
+ /* Ext Playback Switch -> Ext Mono Mixer */
+ {"Ext Mono Right Mixer", NULL, "Ext Right Enable"},
+ {"Ext Mono Right Mixer", "Ext Mono Playback Switch", "Ext Left Enable"},
+ {"Ext Mono Left Mixer", NULL, "Ext Left Enable"},
+ {"Ext Mono Left Mixer", "Ext Mono Playback Switch", "Ext Right Enable"},
+
+ /* HiFi Mono Mixer -> Output Route */
+ {"Earpiece Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Speaker Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Speaker Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+ {"Lineout Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Lineout Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+ {"Headset Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Headset Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+ {"EMU Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"EMU Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+
+ /* Voice PGA -> Output Route */
+ {"Earpiece Playback Route", "Voice", "Voice PGA"},
+ {"Speaker Right Playback Route", "Voice", "Voice PGA"},
+ {"Speaker Left Playback Route", "Voice", "Voice PGA"},
+ {"Lineout Right Playback Route", "Voice", "Voice PGA"},
+ {"Lineout Left Playback Route", "Voice", "Voice PGA"},
+ {"Headset Right Playback Route", "Voice", "Voice PGA"},
+ {"Headset Left Playback Route", "Voice", "Voice PGA"},
+ {"EMU Right Playback Route", "Voice", "Voice PGA"},
+ {"EMU Left Playback Route", "Voice", "Voice PGA"},
+
+ /* Ext Mono Mixer -> Output Route */
+ {"Earpiece Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Speaker Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Speaker Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+ {"Lineout Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Lineout Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+ {"Headset Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Headset Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+ {"EMU Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"EMU Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+
+ /* Output Route -> Output Amplifier */
+ {"Earpiece PGA", NULL, "Earpiece Playback Route"},
+ {"Speaker Right PGA", NULL, "Speaker Right Playback Route"},
+ {"Speaker Left PGA", NULL, "Speaker Left Playback Route"},
+ {"Lineout Right PGA", NULL, "Lineout Right Playback Route"},
+ {"Lineout Left PGA", NULL, "Lineout Left Playback Route"},
+ {"Headset Right PGA", NULL, "Headset Right Playback Route"},
+ {"Headset Left PGA", NULL, "Headset Left Playback Route"},
+ {"EMU Right PGA", NULL, "EMU Right Playback Route"},
+ {"EMU Left PGA", NULL, "EMU Left Playback Route"},
+
+ /* Output Amplifier -> Output */
+ {"EP", NULL, "Earpiece PGA"},
+ {"SPKR", NULL, "Speaker Right PGA"},
+ {"SPKL", NULL, "Speaker Left PGA"},
+ {"LINER", NULL, "Lineout Right PGA"},
+ {"LINEL", NULL, "Lineout Left PGA"},
+ {"HSR", NULL, "Headset Right PGA"},
+ {"HSL", NULL, "Headset Left PGA"},
+ {"EMUR", NULL, "EMU Right PGA"},
+ {"EMUL", NULL, "EMU Left PGA"},
+
+ /* Headset Charge Pump -> Headset */
+ {"HSR", NULL, "Headset Charge Pump"},
+ {"HSL", NULL, "Headset Charge Pump"},
+
+ /* Mic -> Mic Route */
+ {"Right Capture Route", "Mic 1", "MICR"},
+ {"Right Capture Route", "Headset Mic", "HSMIC"},
+ {"Right Capture Route", "EMU Mic", "EMUMIC"},
+ {"Right Capture Route", "Ext Right", "EXTR"},
+ {"Left Capture Route", "Mic 2", "MICL"},
+ {"Left Capture Route", "Ext Left", "EXTL"},
+
+ /* Input Route -> Microphone PGA */
+ {"Microphone 1 PGA", NULL, "Right Capture Route"},
+ {"Microphone 2 PGA", NULL, "Left Capture Route"},
+
+ /* Microphone PGA -> ADC */
+ {"ADC Right", NULL, "Microphone 1 PGA"},
+ {"ADC Left", NULL, "Microphone 2 PGA"},
+
+ /* ADC -> Stream */
+ {"Highpass Filter TX", NULL, "ADC Right"},
+ {"Highpass Filter TX", NULL, "ADC Left"},
+
+ /* Mic Bias */
+ {"MICL", NULL, "MIC1L Bias"},
+ {"MICR", NULL, "MIC1R Bias"},
+};
+
+static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai,
+ int clk_id, int freq)
+{
+ u16 clkfreqreg, clkfreqshift;
+ u16 clkfreqmask, clkfreqval;
+ u16 clkidreg, clkidshift;
+ u16 mask, val;
+ int err;
+
+ switch (dai) {
+ case CPCAP_DAI_HIFI:
+ clkfreqreg = CPCAP_REG_SDAC;
+ clkfreqshift = CPCAP_BIT_ST_DAC_CLK0;
+ clkidreg = CPCAP_REG_SDACDI;
+ clkidshift = CPCAP_BIT_ST_DAC_CLK_IN_SEL;
+ break;
+ case CPCAP_DAI_VOICE:
+ clkfreqreg = CPCAP_REG_CC;
+ clkfreqshift = CPCAP_BIT_CDC_CLK0;
+ clkidreg = CPCAP_REG_CDI;
+ clkidshift = CPCAP_BIT_CLK_IN_SEL;
+ break;
+ default:
+ dev_err(cpcap->codec->dev, "invalid DAI: %d", dai);
+ return -EINVAL;
+ }
+
+ /* setup clk id */
+ if (clk_id < 0 || clk_id > 1) {
+ dev_err(cpcap->codec->dev, "invalid clk id %d", clk_id);
+ return -EINVAL;
+ }
+ err = regmap_update_bits(cpcap->regmap, clkidreg, BIT(clkidshift),
+ clk_id ? BIT(clkidshift) : 0);
+ if (err)
+ return err;
+
+ /* enable PLL for Voice DAI */
+ if (dai == CPCAP_DAI_VOICE) {
+ mask = BIT(CPCAP_BIT_CDC_PLL_SEL);
+ val = BIT(CPCAP_BIT_CDC_PLL_SEL);
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+ mask, val);
+ if (err)
+ return err;
+ }
+
+ /* setup frequency */
+ clkfreqmask = 0x7 << clkfreqshift;
+ switch (freq) {
+ case 15360000:
+ clkfreqval = 0x01 << clkfreqshift;
+ break;
+ case 16800000:
+ clkfreqval = 0x02 << clkfreqshift;
+ break;
+ case 19200000:
+ clkfreqval = 0x03 << clkfreqshift;
+ break;
+ case 26000000:
+ clkfreqval = 0x04 << clkfreqshift;
+ break;
+ case 33600000:
+ clkfreqval = 0x05 << clkfreqshift;
+ break;
+ case 38400000:
+ clkfreqval = 0x06 << clkfreqshift;
+ break;
+ default:
+ dev_err(cpcap->codec->dev, "unsupported freq %u", freq);
+ return -EINVAL;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, clkfreqreg,
+ clkfreqmask, clkfreqval);
+ if (err)
+ return err;
+
+ if (dai == CPCAP_DAI_VOICE) {
+ cpcap->codec_clk_id = clk_id;
+ cpcap->codec_freq = freq;
+ }
+
+ return 0;
+}
+
+static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai,
+ int samplerate)
+{
+ struct snd_soc_codec *codec = cpcap->codec;
+ u16 sampreg, sampmask, sampshift, sampval, sampreset;
+ int err, sampreadval;
+
+ switch (dai) {
+ case CPCAP_DAI_HIFI:
+ sampreg = CPCAP_REG_SDAC;
+ sampshift = CPCAP_BIT_ST_SR0;
+ sampreset = BIT(CPCAP_BIT_DF_RESET_ST_DAC) |
+ BIT(CPCAP_BIT_ST_CLOCK_TREE_RESET);
+ break;
+ case CPCAP_DAI_VOICE:
+ sampreg = CPCAP_REG_CC;
+ sampshift = CPCAP_BIT_CDC_SR0;
+ sampreset = BIT(CPCAP_BIT_DF_RESET) |
+ BIT(CPCAP_BIT_CDC_CLOCK_TREE_RESET);
+ break;
+ default:
+ dev_err(codec->dev, "invalid DAI: %d", dai);
+ return -EINVAL;
+ }
+
+ sampmask = 0xF << sampshift | sampreset;
+ switch (samplerate) {
+ case 48000:
+ sampval = 0x8 << sampshift;
+ break;
+ case 44100:
+ sampval = 0x7 << sampshift;
+ break;
+ case 32000:
+ sampval = 0x6 << sampshift;
+ break;
+ case 24000:
+ sampval = 0x5 << sampshift;
+ break;
+ case 22050:
+ sampval = 0x4 << sampshift;
+ break;
+ case 16000:
+ sampval = 0x3 << sampshift;
+ break;
+ case 12000:
+ sampval = 0x2 << sampshift;
+ break;
+ case 11025:
+ sampval = 0x1 << sampshift;
+ break;
+ case 8000:
+ sampval = 0x0 << sampshift;
+ break;
+ default:
+ dev_err(codec->dev, "unsupported samplerate %d", samplerate);
+ return -EINVAL;
+ }
+ err = regmap_update_bits(cpcap->regmap, sampreg,
+ sampmask, sampval | sampreset);
+ if (err)
+ return err;
+
+ /* Wait for clock tree reset to complete */
+ mdelay(CLOCK_TREE_RESET_TIME);
+
+ err = regmap_read(cpcap->regmap, sampreg, &sampreadval);
+ if (err)
+ return err;
+
+ if (sampreadval & sampreset) {
+ dev_err(codec->dev, "reset self-clear failed: %04x",
+ sampreadval);
+ return -EIO;
+ }
+
+ return 0;
+}
+
+static int cpcap_hifi_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 cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int rate = params_rate(params);
+
+ dev_dbg(codec->dev, "HiFi setup HW params: rate=%d", rate);
+ return cpcap_set_samprate(cpcap, CPCAP_DAI_HIFI, rate);
+}
+
+static int cpcap_hifi_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 cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+
+ dev_dbg(dev, "HiFi setup sysclk: clk_id=%u, freq=%u", clk_id, freq);
+ return cpcap_set_sysclk(cpcap, CPCAP_DAI_HIFI, clk_id, freq);
+}
+
+static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+ static const u16 reg = CPCAP_REG_SDACDI;
+ static const u16 mask =
+ BIT(CPCAP_BIT_SMB_ST_DAC) |
+ BIT(CPCAP_BIT_ST_CLK_INV) |
+ BIT(CPCAP_BIT_ST_FS_INV) |
+ BIT(CPCAP_BIT_ST_DIG_AUD_FS0) |
+ BIT(CPCAP_BIT_ST_DIG_AUD_FS1) |
+ BIT(CPCAP_BIT_ST_L_TIMESLOT0) |
+ BIT(CPCAP_BIT_ST_L_TIMESLOT1) |
+ BIT(CPCAP_BIT_ST_L_TIMESLOT2) |
+ BIT(CPCAP_BIT_ST_R_TIMESLOT0) |
+ BIT(CPCAP_BIT_ST_R_TIMESLOT1) |
+ BIT(CPCAP_BIT_ST_R_TIMESLOT2);
+ u16 val = 0x0000;
+
+ dev_dbg(dev, "HiFi setup dai format (%08x)", fmt);
+
+ /*
+ * "HiFi Playback" should always be configured as
+ * SND_SOC_DAIFMT_CBM_CFM - codec clk & frm master
+ * SND_SOC_DAIFMT_I2S - I2S mode
+ */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val &= ~BIT(CPCAP_BIT_SMB_ST_DAC);
+ break;
+ default:
+ dev_err(dev, "HiFi dai fmt failed: CPCAP should be master");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ val |= BIT(CPCAP_BIT_ST_FS_INV);
+ val |= BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ val &= ~BIT(CPCAP_BIT_ST_FS_INV);
+ val |= BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ val |= BIT(CPCAP_BIT_ST_FS_INV);
+ val &= ~BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ val &= ~BIT(CPCAP_BIT_ST_FS_INV);
+ val &= ~BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ default:
+ dev_err(dev, "HiFi dai fmt failed: unsupported clock invert mode");
+ return -EINVAL;
+ }
+
+ if (val & BIT(CPCAP_BIT_ST_CLK_INV))
+ val &= ~BIT(CPCAP_BIT_ST_CLK_INV);
+ else
+ val |= BIT(CPCAP_BIT_ST_CLK_INV);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ val |= BIT(CPCAP_BIT_ST_DIG_AUD_FS0);
+ val |= BIT(CPCAP_BIT_ST_DIG_AUD_FS1);
+ break;
+ default:
+ /* 01 - 4 slots network mode */
+ val |= BIT(CPCAP_BIT_ST_DIG_AUD_FS0);
+ val &= ~BIT(CPCAP_BIT_ST_DIG_AUD_FS1);
+ /* L on slot 1 */
+ val |= BIT(CPCAP_BIT_ST_L_TIMESLOT0);
+ break;
+ }
+
+ dev_dbg(dev, "HiFi dai format: val=%04x", val);
+ return regmap_update_bits(cpcap->regmap, reg, mask, val);
+}
+
+static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 reg = CPCAP_REG_RXSDOA;
+ static const u16 mask = BIT(CPCAP_BIT_ST_DAC_SW);
+ u16 val;
+
+ if (mute)
+ val = 0;
+ else
+ val = BIT(CPCAP_BIT_ST_DAC_SW);
+
+ dev_dbg(codec->dev, "HiFi mute: %d", mute);
+ return regmap_update_bits(cpcap->regmap, reg, mask, val);
+}
+
+static const struct snd_soc_dai_ops cpcap_dai_hifi_ops = {
+ .hw_params = cpcap_hifi_hw_params,
+ .set_sysclk = cpcap_hifi_set_dai_sysclk,
+ .set_fmt = cpcap_hifi_set_dai_fmt,
+ .digital_mute = cpcap_hifi_set_mute,
+};
+
+static int cpcap_voice_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 device *dev = codec->dev;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 reg_cdi = CPCAP_REG_CDI;
+ int rate = params_rate(params);
+ int channels = params_channels(params);
+ int direction = substream->stream;
+ u16 val, mask;
+ int err;
+
+ dev_dbg(dev, "Voice setup HW params: rate=%d, direction=%d, chan=%d",
+ rate, direction, channels);
+
+ err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, rate);
+ if (err)
+ return err;
+
+ if (direction == SNDRV_PCM_STREAM_CAPTURE) {
+ mask = 0x0000;
+ mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0;
+ mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1;
+ mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2;
+ mask |= CPCAP_BIT_MIC2_TIMESLOT0;
+ mask |= CPCAP_BIT_MIC2_TIMESLOT1;
+ mask |= CPCAP_BIT_MIC2_TIMESLOT2;
+ val = 0x0000;
+ if (channels >= 2)
+ val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0);
+ err = regmap_update_bits(cpcap->regmap, reg_cdi, mask, val);
+ if (err)
+ return err;
+ }
+
+ return 0;
+}
+
+static int cpcap_voice_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 cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "Voice setup sysclk: clk_id=%u, freq=%u",
+ clk_id, freq);
+ return cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, clk_id, freq);
+}
+
+static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 mask = BIT(CPCAP_BIT_SMB_CDC) |
+ BIT(CPCAP_BIT_CLK_INV) |
+ BIT(CPCAP_BIT_FS_INV) |
+ BIT(CPCAP_BIT_CDC_DIG_AUD_FS0) |
+ BIT(CPCAP_BIT_CDC_DIG_AUD_FS1);
+ u16 val = 0x0000;
+ int err;
+
+ dev_dbg(codec->dev, "Voice setup dai format (%08x)", fmt);
+
+ /*
+ * "Voice Playback" and "Voice Capture" should always be
+ * configured as SND_SOC_DAIFMT_CBM_CFM - codec clk & frm
+ * master
+ */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val &= ~BIT(CPCAP_BIT_SMB_CDC);
+ break;
+ default:
+ dev_err(codec->dev, "Voice dai fmt failed: CPCAP should be the master");
+ val &= ~BIT(CPCAP_BIT_SMB_CDC);
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ val |= BIT(CPCAP_BIT_CLK_INV);
+ val |= BIT(CPCAP_BIT_FS_INV);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ val |= BIT(CPCAP_BIT_CLK_INV);
+ val &= ~BIT(CPCAP_BIT_FS_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ val &= ~BIT(CPCAP_BIT_CLK_INV);
+ val |= BIT(CPCAP_BIT_FS_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ val &= ~BIT(CPCAP_BIT_CLK_INV);
+ val &= ~BIT(CPCAP_BIT_FS_INV);
+ break;
+ default:
+ dev_err(codec->dev, "Voice dai fmt failed: unsupported clock invert mode");
+ break;
+ }
+
+ if (val & BIT(CPCAP_BIT_CLK_INV))
+ val &= ~BIT(CPCAP_BIT_CLK_INV);
+ else
+ val |= BIT(CPCAP_BIT_CLK_INV);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* 11 - true I2S mode */
+ val |= BIT(CPCAP_BIT_CDC_DIG_AUD_FS0);
+ val |= BIT(CPCAP_BIT_CDC_DIG_AUD_FS1);
+ break;
+ default:
+ /* 4 timeslots network mode */
+ val |= BIT(CPCAP_BIT_CDC_DIG_AUD_FS0);
+ val &= ~BIT(CPCAP_BIT_CDC_DIG_AUD_FS1);
+ break;
+ }
+
+ dev_dbg(codec->dev, "Voice dai format: val=%04x", val);
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, mask, val);
+ if (err)
+ return err;
+
+ cpcap->codec_format = val;
+ return 0;
+}
+
+static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 reg = CPCAP_REG_RXCOA;
+ static const u16 mask = BIT(CPCAP_BIT_CDC_SW);
+ u16 val;
+
+ if (mute)
+ val = 0;
+ else
+ val = BIT(CPCAP_BIT_CDC_SW);
+
+ dev_dbg(codec->dev, "Voice mute: %d", mute);
+ return regmap_update_bits(cpcap->regmap, reg, mask, val);
+};
+
+static const struct snd_soc_dai_ops cpcap_dai_voice_ops = {
+ .hw_params = cpcap_voice_hw_params,
+ .set_sysclk = cpcap_voice_set_dai_sysclk,
+ .set_fmt = cpcap_voice_set_dai_fmt,
+ .digital_mute = cpcap_voice_set_mute,
+};
+
+static struct snd_soc_dai_driver cpcap_dai[] = {
+{
+ .id = 0,
+ .name = "cpcap-hifi",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE,
+ },
+ .ops = &cpcap_dai_hifi_ops,
+},
+{
+ .id = 1,
+ .name = "cpcap-voice",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Voice Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &cpcap_dai_voice_ops,
+},
+};
+
+static int cpcap_dai_mux(struct cpcap_audio *cpcap, bool swap_dai_configuration)
+{
+ u16 hifi_val, voice_val;
+ u16 hifi_mask = BIT(CPCAP_BIT_DIG_AUD_IN_ST_DAC);
+ u16 voice_mask = BIT(CPCAP_BIT_DIG_AUD_IN);
+ int err;
+
+
+
+ if (!swap_dai_configuration) {
+ /* Codec on DAI0, HiFi on DAI1 */
+ voice_val = 0;
+ hifi_val = hifi_mask;
+ } else {
+ /* Codec on DAI1, HiFi on DAI0 */
+ voice_val = voice_mask;
+ hifi_val = 0;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+ voice_mask, voice_val);
+ if (err)
+ return err;
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_SDACDI,
+ hifi_mask, hifi_val);
+ if (err)
+ return err;
+
+ return 0;
+}
+
+static int cpcap_audio_reset(struct snd_soc_codec *codec,
+ bool swap_dai_configuration)
+{
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int i, err = 0;
+
+ dev_dbg(codec->dev, "init audio codec");
+
+ for (i = 0; i < ARRAY_SIZE(cpcap_default_regs); i++) {
+ err = regmap_update_bits(cpcap->regmap,
+ cpcap_default_regs[i].reg,
+ cpcap_default_regs[i].mask,
+ cpcap_default_regs[i].val);
+ if (err)
+ return err;
+ }
+
+ /* setup default settings */
+ err = cpcap_dai_mux(cpcap, swap_dai_configuration);
+ if (err)
+ return err;
+
+ err = cpcap_set_sysclk(cpcap, CPCAP_DAI_HIFI, 0, 26000000);
+ if (err)
+ return err;
+ err = cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, 0, 26000000);
+ if (err)
+ return err;
+
+ err = cpcap_set_samprate(cpcap, CPCAP_DAI_HIFI, 48000);
+ if (err)
+ return err;
+
+ err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, 48000);
+ if (err)
+ return err;
+
+ return 0;
+}
+
+static int cpcap_soc_probe(struct snd_soc_codec *codec)
+{
+ struct cpcap_audio *cpcap;
+ int err;
+
+ cpcap = devm_kzalloc(codec->dev, sizeof(*cpcap), GFP_KERNEL);
+ if (!cpcap)
+ return -ENOMEM;
+ snd_soc_codec_set_drvdata(codec, cpcap);
+ cpcap->codec = codec;
+
+ cpcap->regmap = dev_get_regmap(codec->dev->parent, NULL);
+ if (!cpcap->regmap)
+ return -ENODEV;
+ snd_soc_codec_init_regmap(codec, cpcap->regmap);
+
+ err = cpcap_get_vendor(codec->dev, cpcap->regmap, &cpcap->vendor);
+ if (err)
+ return err;
+
+ return cpcap_audio_reset(codec, false);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cpcap = {
+ .probe = cpcap_soc_probe,
+
+ .component_driver = {
+ .controls = cpcap_snd_controls,
+ .num_controls = ARRAY_SIZE(cpcap_snd_controls),
+ .dapm_widgets = cpcap_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cpcap_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
+ },
+};
+
+static int cpcap_codec_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cpcap,
+ cpcap_dai, ARRAY_SIZE(cpcap_dai));
+}
+
+static int cpcap_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id cpcap_audio_of_match[] = {
+ { .compatible = "motorola,cpcap-audio-codec", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, cpcap_audio_of_match);
+#endif
+
+static struct platform_driver cpcap_codec_driver = {
+ .probe = cpcap_codec_probe,
+ .remove = cpcap_codec_remove,
+ .driver = {
+ .name = "cpcap-codec",
+ .of_match_table = of_match_ptr(cpcap_audio_of_match),
+ },
+};
+module_platform_driver(cpcap_codec_driver);
+
+MODULE_ALIAS("platform:cpcap-codec");
+MODULE_DESCRIPTION("ASoC CPCAP codec driver");
+MODULE_AUTHOR("Sebastian Reichel");
+MODULE_LICENSE("GPL");
--
2.15.1
Motorola CPCAP is a PMIC with audio functionality, that can be
found on Motorola Droid 4 and probably a few other phones from
Motorola's Droid series.
This adds the DT binding for the codec sub-module found inside
the PMIC.
Acked-by: Rob Herring <[email protected]>
Signed-off-by: Sebastian Reichel <[email protected]>
---
.../bindings/sound/motorola,cpcap-audio-codec.txt | 17 +++++++++++++++++
1 file changed, 17 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
diff --git a/Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt b/Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
new file mode 100644
index 000000000000..a6b64132b545
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
@@ -0,0 +1,17 @@
+Motorola CPCAP audio CODEC
+--------------------------
+
+This module is part of the CPCAP. For more details about the whole
+chip see Documentation/devicetree/bindings/mfd/motorola-cpcap.txt.
+
+Required properties:
+
+ - compatible : "motorola,cpcap-audio-codec"
+
+Example:
+
+&cpcap {
+ audio-codec {
+ compatible = "motorola,cpcap-audio-codec";
+ };
+};
--
2.15.1
Dear Sebastian,
On Wed, Feb 14, 2018 at 11:07 PM, Sebastian Reichel
<[email protected]> wrote:
> Motorola CPCAP is a PMIC with audio functionality, that can be
> found on Motorola Droid 4 and probably a few other phones from
> Motorola's Droid series.
>
> The driver has been written from scratch using Motorola's Android
> driver, register dumps from running Android and datasheet for NXP
> MC13783UG (which is similar to Motorola CPCAP, but not the same).
>
> The chip provides two audio interfaces, that can be muxed to two
> different audio codecs. One provides support for stereo output
> (named StDAC or HiFi), while the other only provides mono output
> (named Voice). Only the Voice codec provides a Capture interface.
>
> Signed-off-by: Sebastian Reichel <[email protected]>
<snip>
> --- /dev/null
> +++ b/sound/soc/codecs/cpcap.c
> @@ -0,0 +1,1588 @@
> +/*
> + * ALSA SoC CPCAP codec driver
> + *
> + * Copyright (C) 2017 - 2018 Sebastian Reichel <[email protected]>
> + *
> + * Very loosely based on original driver from Motorola:
> + * Copyright (C) 2007 - 2009 Motorola, Inc.
> + *
> + * 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.
> + *
> + */
Could you consider using the new SPDX tags instead of this fine but
long legalese? [1]
<snip>
> +MODULE_ALIAS("platform:cpcap-codec");
> +MODULE_DESCRIPTION("ASoC CPCAP codec driver");
> +MODULE_AUTHOR("Sebastian Reichel");
> +MODULE_LICENSE("GPL");
This does not match your license: per module.h, GPL means GPL-2.0 or
later. You notice above is GPL-2.0, not "or later"
[1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Documentation/process/license-rules.rst
--
Cordially
Philippe Ombredanne
On Wed, Feb 14, 2018 at 11:07:39PM +0100, Sebastian Reichel wrote:
> +config SND_SOC_CPCAP
> + tristate "Motorola CPCAP codec"
> + depends on MFD_CPCAP
> + default MFD_CPCAP
> +
We don't have default lines like this for other MFDs...
> + SOC_ENUM("Hifi Left Phase Playback Switch",
> + cpcap_hifi_l_phase_inv_enum),
> + SOC_ENUM("Ext Left Phase Playback Switch",
> + cpcap_ext_l_phase_inv_enum),
If this is a Switch control it should be a simple on/off binary control.
If it's an enumeration it shouldn't have Switch at the end of the name
since that's a keyword interpreted by userspace. Either option seems
fine in this case, if it's a switch it should be "Foo Invert Switch" or
similar.
On Wed, Feb 14, 2018 at 11:07:38PM +0100, Sebastian Reichel wrote:
> +&cpcap {
> + audio-codec {
> + compatible = "motorola,cpcap-audio-codec";
> + };
Why are we adding a separate DT node with no content for this? This is
a single chip, we already know that the CODEC part is there from the DT
telling us that the chip is there and what we decide is part of the
CODEC is going to depend on what the OS running on the system is doing.
Hi Philippe,
On Thu, Feb 15, 2018 at 10:50:22AM +0100, Philippe Ombredanne wrote:
> Dear Sebastian,
>
> On Wed, Feb 14, 2018 at 11:07 PM, Sebastian Reichel
> <[email protected]> wrote:
> > Motorola CPCAP is a PMIC with audio functionality, that can be
> > found on Motorola Droid 4 and probably a few other phones from
> > Motorola's Droid series.
> >
> > The driver has been written from scratch using Motorola's Android
> > driver, register dumps from running Android and datasheet for NXP
> > MC13783UG (which is similar to Motorola CPCAP, but not the same).
> >
> > The chip provides two audio interfaces, that can be muxed to two
> > different audio codecs. One provides support for stereo output
> > (named StDAC or HiFi), while the other only provides mono output
> > (named Voice). Only the Voice codec provides a Capture interface.
> >
> > Signed-off-by: Sebastian Reichel <[email protected]>
>
> <snip>
>
> > --- /dev/null
> > +++ b/sound/soc/codecs/cpcap.c
> > @@ -0,0 +1,1588 @@
> > +/*
> > + * ALSA SoC CPCAP codec driver
> > + *
> > + * Copyright (C) 2017 - 2018 Sebastian Reichel <[email protected]>
> > + *
> > + * Very loosely based on original driver from Motorola:
> > + * Copyright (C) 2007 - 2009 Motorola, Inc.
> > + *
> > + * 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.
> > + *
> > + */
>
> Could you consider using the new SPDX tags instead of this fine but
> long legalese? [1]
Sure. Thanks for the hint.
> <snip>
>
> > +MODULE_ALIAS("platform:cpcap-codec");
> > +MODULE_DESCRIPTION("ASoC CPCAP codec driver");
> > +MODULE_AUTHOR("Sebastian Reichel");
> > +MODULE_LICENSE("GPL");
>
> This does not match your license: per module.h, GPL means GPL-2.0 or
> later. You notice above is GPL-2.0, not "or later"
Indeed. I will fix this in the next revision.
> [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Documentation/process/license-rules.rst
-- Sebastian
Hi Mark,
On Fri, Feb 16, 2018 at 11:30:08AM +0000, Mark Brown wrote:
> On Wed, Feb 14, 2018 at 11:07:38PM +0100, Sebastian Reichel wrote:
>
> > +&cpcap {
> > + audio-codec {
> > + compatible = "motorola,cpcap-audio-codec";
> > + };
>
> Why are we adding a separate DT node with no content for this? This is
> a single chip, we already know that the CODEC part is there from the DT
> telling us that the chip is there and what we decide is part of the
> CODEC is going to depend on what the OS running on the system is doing.
While it looks empty in the DT binding file, it's actually not empty
once some standard properties are added to support audio-graph-card.
A real world example looks like this:
&cpcap {
audio-codec {
compatible = "motorola,cpcap-audio-codec";
#sound-dai-cells = <1>;
port@0 {
cpcap_audio_codec0: endpoint {
remote-endpoint = <&cpu_dai2>;
};
};
port@1 {
cpcap_audio_codec1: endpoint {
remote-endpoint = <&cpu_dai3>;
};
};
};
};
Having all of this directly in the cpcap node doesn't look like
an improvement for any OS. Once there is a node it makes sense
to add a compatible IMHO. As a side effect this also avoids having
special handling for the audio codec. This is the last missing
sub-component, see arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
-- Sebastian
On Fri, Feb 16, 2018 at 02:25:38PM +0100, Sebastian Reichel wrote:
> On Fri, Feb 16, 2018 at 11:30:08AM +0000, Mark Brown wrote:
> > Why are we adding a separate DT node with no content for this? This is
> > a single chip, we already know that the CODEC part is there from the DT
> > telling us that the chip is there and what we decide is part of the
> > CODEC is going to depend on what the OS running on the system is doing.
> While it looks empty in the DT binding file, it's actually not empty
> once some standard properties are added to support audio-graph-card.
This tells me you're missing something in the binding defining the DAIs
and...
> A real world example looks like this:
> &cpcap {
> audio-codec {
> compatible = "motorola,cpcap-audio-codec";
> #sound-dai-cells = <1>;
...that still doesn't require a compatible here.
Hi Mark,
On Fri, Feb 16, 2018 at 11:39:01AM +0000, Mark Brown wrote:
> On Wed, Feb 14, 2018 at 11:07:39PM +0100, Sebastian Reichel wrote:
>
> > +config SND_SOC_CPCAP
> > + tristate "Motorola CPCAP codec"
> > + depends on MFD_CPCAP
> > + default MFD_CPCAP
> > +
>
> We don't have default lines like this for other MFDs...
Is there a reason not to add them? It looks very useful to me:
99% of people are not interested in the MFD. They don't enable
it and are not affected at all. They do not even see extra
configuration entry due to the dependency.
0.9% of people have the MFD in their system and want their
hardware to 'just work'. With this line it's enough to enable
the MFD and the audio subsystem, so less knowledge and configuration
is required.
0.1% of people have special requirements and want the MFD
and generic audio support, but no support for the MFD audio
codec. There is a high chance, that they know how to configure
it properly :)
> > + SOC_ENUM("Hifi Left Phase Playback Switch",
> > + cpcap_hifi_l_phase_inv_enum),
> > + SOC_ENUM("Ext Left Phase Playback Switch",
> > + cpcap_ext_l_phase_inv_enum),
>
> If this is a Switch control it should be a simple on/off binary control.
> If it's an enumeration it shouldn't have Switch at the end of the name
> since that's a keyword interpreted by userspace. Either option seems
> fine in this case, if it's a switch it should be "Foo Invert Switch" or
> similar.
Ok.
-- Sebastian
Hi,
On Fri, Feb 16, 2018 at 01:44:48PM +0000, Mark Brown wrote:
> On Fri, Feb 16, 2018 at 02:25:38PM +0100, Sebastian Reichel wrote:
> > On Fri, Feb 16, 2018 at 11:30:08AM +0000, Mark Brown wrote:
>
> > > Why are we adding a separate DT node with no content for this? This is
> > > a single chip, we already know that the CODEC part is there from the DT
> > > telling us that the chip is there and what we decide is part of the
> > > CODEC is going to depend on what the OS running on the system is doing.
>
> > While it looks empty in the DT binding file, it's actually not empty
> > once some standard properties are added to support audio-graph-card.
>
> This tells me you're missing something in the binding defining the
> DAIs and...
Well it is described by the following document:
Documentation/devicetree/bindings/sound/audio-graph-card.txt
Previous revision of the codec also worked perfectly fine
with the simple card binding, which does the DAI stuff
differently:
Documentation/devicetree/bindings/sound/simple-card.txt
A quick check of the other codecs suggested, that none of them
descibes the graph based binding style. AFAIUI they could be
used with it, though. So if you have a suggestion for a better
binding document I can adopt this in the next version.
> > A real world example looks like this:
>
> > &cpcap {
> > audio-codec {
> > compatible = "motorola,cpcap-audio-codec";
> > #sound-dai-cells = <1>;
>
> ...that still doesn't require a compatible here.
I agree, that it's not required. Also the node is not required.
Everything could be dumped into the main node. Many things are
not required, but they make implementations easier and help in
regards to DT readability and consistency. Having the compatible
means, that all sub-functions _can_ be handled equally by the
operating system. Not having the compatible means you _always_
need special handling for the audio codec. This basically makes
the codec node different for the simple purpose of "because it is
not strictly required". If we have a compatible node, other
operating systems can still decide to ignore it, right?
-- Sebastian
On Fri, Feb 16, 2018 at 02:51:13PM +0100, Sebastian Reichel wrote:
> On Fri, Feb 16, 2018 at 11:39:01AM +0000, Mark Brown wrote:
> > We don't have default lines like this for other MFDs...
> Is there a reason not to add them? It looks very useful to me:
Consistency. It's a lot easier to apply a single rule to everything
than it is to have to individually think through and discuss the
decisions on every individual driver, especially when there will
inevitably be some cases where there are conflicting requirements from
different users. As you said in the rest of your mail hand configuring
kernels is already a very specialist thing at the best of times.
On Fri, Feb 16, 2018 at 03:12:37PM +0100, Sebastian Reichel wrote:
> On Fri, Feb 16, 2018 at 01:44:48PM +0000, Mark Brown wrote:
> > On Fri, Feb 16, 2018 at 02:25:38PM +0100, Sebastian Reichel wrote:
> > > While it looks empty in the DT binding file, it's actually not empty
> > > once some standard properties are added to support audio-graph-card.
> > This tells me you're missing something in the binding defining the
> > DAIs and...
> Well it is described by the following document:
> Documentation/devicetree/bindings/sound/audio-graph-card.txt
You still need to say which DAIs exist on the device and how they are
identified - if there's only one DAI it's obviously easy but if a device
has multiple DAIs then there's some naming to do.
> > ...that still doesn't require a compatible here.
> I agree, that it's not required. Also the node is not required.
> Everything could be dumped into the main node. Many things are
> not required, but they make implementations easier and help in
> regards to DT readability and consistency. Having the compatible
> means, that all sub-functions _can_ be handled equally by the
> operating system. Not having the compatible means you _always_
> need special handling for the audio codec. This basically makes
> the codec node different for the simple purpose of "because it is
> not strictly required". If we have a compatible node, other
> operating systems can still decide to ignore it, right?
It's not just other operating systems, it's also other versions of
Linux we have to think about here. The most obvious issue with audio is
the clocking where the division between ASoC and clock APIs is not super
obvious and could easily change in the future.
* Mark Brown <[email protected]> [180216 15:17]:
> On Fri, Feb 16, 2018 at 03:12:37PM +0100, Sebastian Reichel wrote:
> > On Fri, Feb 16, 2018 at 01:44:48PM +0000, Mark Brown wrote:
> > > On Fri, Feb 16, 2018 at 02:25:38PM +0100, Sebastian Reichel wrote:
>
> > > > While it looks empty in the DT binding file, it's actually not empty
> > > > once some standard properties are added to support audio-graph-card.
>
> > > This tells me you're missing something in the binding defining the
> > > DAIs and...
>
> > Well it is described by the following document:
>
> > Documentation/devicetree/bindings/sound/audio-graph-card.txt
>
> You still need to say which DAIs exist on the device and how they are
> identified - if there's only one DAI it's obviously easy but if a device
> has multiple DAIs then there's some naming to do.
>
> > > ...that still doesn't require a compatible here.
>
> > I agree, that it's not required. Also the node is not required.
> > Everything could be dumped into the main node. Many things are
> > not required, but they make implementations easier and help in
> > regards to DT readability and consistency. Having the compatible
> > means, that all sub-functions _can_ be handled equally by the
> > operating system. Not having the compatible means you _always_
> > need special handling for the audio codec. This basically makes
> > the codec node different for the simple purpose of "because it is
> > not strictly required". If we have a compatible node, other
> > operating systems can still decide to ignore it, right?
>
> It's not just other operating systems, it's also other versions of
> Linux we have to think about here. The most obvious issue with audio is
> the clocking where the division between ASoC and clock APIs is not super
> obvious and could easily change in the future.
Yeah let's stick to describing how the hardware is wired in the
dts files. In this case it's really only the routing to the SoC,
right?
One advantage of using a compatible property for the pmic subdevices
though is that it leaves out a dependency between various device
drivers things happen automagically. The mfd core driver can be
minimal and just implement interrupt handling and regmap. So no need
to to parse the child nodes in the pmic mfd driver :)
So personally I'd prefer the option that requires least amount
of custom code if compatible vs no compatible property is the
only issue here.
Regards,
Tony
Hi,
On Fri, Feb 16, 2018 at 03:16:09PM +0000, Mark Brown wrote:
> On Fri, Feb 16, 2018 at 03:12:37PM +0100, Sebastian Reichel wrote:
> > On Fri, Feb 16, 2018 at 01:44:48PM +0000, Mark Brown wrote:
> > > On Fri, Feb 16, 2018 at 02:25:38PM +0100, Sebastian Reichel wrote:
>
> > > > While it looks empty in the DT binding file, it's actually not empty
> > > > once some standard properties are added to support audio-graph-card.
>
> > > This tells me you're missing something in the binding defining the
> > > DAIs and...
>
> > Well it is described by the following document:
>
> > Documentation/devicetree/bindings/sound/audio-graph-card.txt
>
> You still need to say which DAIs exist on the device and how they are
> identified - if there's only one DAI it's obviously easy but if a device
> has multiple DAIs then there's some naming to do.
Ok, I will add some more description of the codec's capabilities.
> > > ...that still doesn't require a compatible here.
>
> > I agree, that it's not required. Also the node is not required.
> > Everything could be dumped into the main node. Many things are
> > not required, but they make implementations easier and help in
> > regards to DT readability and consistency. Having the compatible
> > means, that all sub-functions _can_ be handled equally by the
> > operating system. Not having the compatible means you _always_
> > need special handling for the audio codec. This basically makes
> > the codec node different for the simple purpose of "because it is
> > not strictly required". If we have a compatible node, other
> > operating systems can still decide to ignore it, right?
>
> It's not just other operating systems, it's also other versions of
> Linux we have to think about here. The most obvious issue with audio is
> the clocking where the division between ASoC and clock APIs is not super
> obvious and could easily change in the future.
Which does not change my argument. Other handling in Linux is
basically equal to other operating system. We don't loose anything
by having a compatible available.
-- Sebastian
On Fri, Feb 16, 2018 at 07:57:07AM -0800, Tony Lindgren wrote:
> * Mark Brown <[email protected]> [180216 15:17]:
> > It's not just other operating systems, it's also other versions of
> > Linux we have to think about here. The most obvious issue with audio is
> > the clocking where the division between ASoC and clock APIs is not super
> > obvious and could easily change in the future.
> Yeah let's stick to describing how the hardware is wired in the
> dts files. In this case it's really only the routing to the SoC,
> right?
The SoC and anything else on the board like external amps and so on.
> One advantage of using a compatible property for the pmic subdevices
> though is that it leaves out a dependency between various device
> drivers things happen automagically. The mfd core driver can be
> minimal and just implement interrupt handling and regmap. So no need
> to to parse the child nodes in the pmic mfd driver :)
There's no need to do that anyway - with a MFD the child devices can
assume that they're part of the MFD and reference their parent.
> So personally I'd prefer the option that requires least amount
> of custom code if compatible vs no compatible property is the
> only issue here.
It's a few lines of code to register the child devices from code rather
than the DT, and keeps it out of the ABI.
On Fri, Feb 16, 2018 at 04:58:37PM +0100, Sebastian Reichel wrote:
> On Fri, Feb 16, 2018 at 03:16:09PM +0000, Mark Brown wrote:
> > It's not just other operating systems, it's also other versions of
> > Linux we have to think about here. The most obvious issue with audio is
> > the clocking where the division between ASoC and clock APIs is not super
> > obvious and could easily change in the future.
> Which does not change my argument. Other handling in Linux is
> basically equal to other operating system. We don't loose anything
> by having a compatible available.
It's bad practice to do it - we need to remember we're designing ABIs
here and try to do the best job we can of them.
* Mark Brown <[email protected]> [180219 12:05]:
> On Fri, Feb 16, 2018 at 07:57:07AM -0800, Tony Lindgren wrote:
> > One advantage of using a compatible property for the pmic subdevices
> > though is that it leaves out a dependency between various device
> > drivers things happen automagically. The mfd core driver can be
> > minimal and just implement interrupt handling and regmap. So no need
> > to to parse the child nodes in the pmic mfd driver :)
>
> There's no need to do that anyway - with a MFD the child devices can
> assume that they're part of the MFD and reference their parent.
>
> > So personally I'd prefer the option that requires least amount
> > of custom code if compatible vs no compatible property is the
> > only issue here.
>
> It's a few lines of code to register the child devices from code rather
> than the DT, and keeps it out of the ABI.
OK yeah that's a good point with avoiding the ABI. Seems
we still want the dts child node(s) though. That way audio
device can be disabled for devices where audio is not wired
up at all on this PMIC.
Regards,
Tony
On Fri 2018-02-16 14:27:16, Mark Brown wrote:
> On Fri, Feb 16, 2018 at 02:51:13PM +0100, Sebastian Reichel wrote:
> > On Fri, Feb 16, 2018 at 11:39:01AM +0000, Mark Brown wrote:
>
> > > We don't have default lines like this for other MFDs...
>
> > Is there a reason not to add them? It looks very useful to me:
>
> Consistency. It's a lot easier to apply a single rule to everything
> than it is to have to individually think through and discuss the
> decisions on every individual driver, especially when there will
> inevitably be some cases where there are conflicting requirements from
> different users. As you said in the rest of your mail hand configuring
> kernels is already a very specialist thing at the best of times.
Yes, so please don't make it any harder than it needs to be. Having to
enable (by hand) more than one option per chip is annoying and error-prone.
Pavel
--
(english) http://www.livejournal.com/~pavelmachek
(cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html
Hi,
On Fri, Feb 23, 2018 at 09:07:14AM +0100, Pavel Machek wrote:
> On Fri 2018-02-16 14:27:16, Mark Brown wrote:
> > On Fri, Feb 16, 2018 at 02:51:13PM +0100, Sebastian Reichel wrote:
> > > On Fri, Feb 16, 2018 at 11:39:01AM +0000, Mark Brown wrote:
> >
> > > > We don't have default lines like this for other MFDs...
> >
> > > Is there a reason not to add them? It looks very useful to me:
> >
> > Consistency. It's a lot easier to apply a single rule to everything
> > than it is to have to individually think through and discuss the
> > decisions on every individual driver, especially when there will
> > inevitably be some cases where there are conflicting requirements from
> > different users. As you said in the rest of your mail hand configuring
> > kernels is already a very specialist thing at the best of times.
>
> Yes, so please don't make it any harder than it needs to be. Having to
> enable (by hand) more than one option per chip is annoying and error-prone.
I will drop this and instead provide a new patch, which converts all
codecs depending on MFD_XYZ to have "default MFD_XYZ". This means,
that there will be consistency and we can discuss this with a bigger
audience without blocking the actual driver patches.
-- Sebastian
Hi,
On Thu, Feb 22, 2018 at 11:54:19AM -0800, Tony Lindgren wrote:
> * Mark Brown <[email protected]> [180219 12:05]:
> > On Fri, Feb 16, 2018 at 07:57:07AM -0800, Tony Lindgren wrote:
> > > One advantage of using a compatible property for the pmic subdevices
> > > though is that it leaves out a dependency between various device
> > > drivers things happen automagically. The mfd core driver can be
> > > minimal and just implement interrupt handling and regmap. So no need
> > > to to parse the child nodes in the pmic mfd driver :)
> >
> > There's no need to do that anyway - with a MFD the child devices can
> > assume that they're part of the MFD and reference their parent.
> >
> > > So personally I'd prefer the option that requires least amount
> > > of custom code if compatible vs no compatible property is the
> > > only issue here.
> >
> > It's a few lines of code to register the child devices from code rather
> > than the DT, and keeps it out of the ABI.
>
> OK yeah that's a good point with avoiding the ABI. Seems
> we still want the dts child node(s) though. That way audio
> device can be disabled for devices where audio is not wired
> up at all on this PMIC.
We need something to identify the correct child node. If there
is no compatible, the node name will become ABI. So I don't
think we gain anything by removing the compatible. As far as
I can see it seems to be unusual to use fixed node names. I
could find some examples, but most sub-devices are identified
using compatibles. This method also has direct support in MFD
(using .of_compatible in mfd_cell).
Based on a bit of grepping through drivers/mfd/, the node name
based identification seems to be mainly used by ASoC, while most
other subsystems seem to prefer compatible based identification.
Anyways, I have implemented the node name based identification and
will post patches once I fixed the enum issue. I will probably
send them later today.
-- Sebastian