[PATCH 0/3] Motorola Droid 4 Audio Support

[PATCH 0/3] Motorola Droid 4 Audio Support

[Sebastian Reichel]

Hi,

I got working sound on Droid 4 with mainline \o/. The codec is
currently missing support for detecting if something has been
plugged into the 3.5mm connector, since that seems to require
some closed source firmware and needs further investigation. I
think this can be added later.

Tested:
 - Playing via [HiFi] using Speaker, Earpiece, Headphone
 - Playing via [Voice] using Speaker, Earpiece, Headphone
 - Recording via [Voice] using Mic
 - Volume Controls
 - Checking that DAPM graph look correct

-- Sebastian

Sebastian Reichel (3):
  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  |   19 +
 arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi     |    6 +
 arch/arm/boot/dts/omap4-droid4-xt894.dts           |   76 ++
 sound/soc/codecs/Kconfig                           |    3 +
 sound/soc/codecs/Makefile                          |    2 +
 sound/soc/codecs/cpcap.c                           | 1422 ++++++++++++++++++++
 6 files changed, 1528 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
 create mode 100644 sound/soc/codecs/cpcap.c

-- 
2.13.2

[PATCH 0/3] Motorola Droid 4 Audio Support

[Sebastian Reichel]

Hi,

I got working sound on Droid 4 with mainline \o/. The codec is
currently missing support for detecting if something has been
plugged into the 3.5mm connector, since that seems to require
some closed source firmware and needs further investigation. I
think this can be added later.

Tested:
 - Playing via [HiFi] using Speaker, Earpiece, Headphone
 - Playing via [Voice] using Speaker, Earpiece, Headphone
 - Recording via [Voice] using Mic
 - Volume Controls
 - Checking that DAPM graph look correct

-- Sebastian

Sebastian Reichel (3):
  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  |   19 +
 arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi     |    6 +
 arch/arm/boot/dts/omap4-droid4-xt894.dts           |   76 ++
 sound/soc/codecs/Kconfig                           |    3 +
 sound/soc/codecs/Makefile                          |    2 +
 sound/soc/codecs/cpcap.c                           | 1422 ++++++++++++++++++++
 6 files changed, 1528 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
 create mode 100644 sound/soc/codecs/cpcap.c

-- 
2.13.2

[PATCH 1/3] ASoC: codec: cpcap: new codec

[Sebastian Reichel]

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 <sebastian.reichel@collabora.co.uk>
---
 .../bindings/sound/motorola,cpcap-audio-codec.txt  |   19 +
 sound/soc/codecs/Kconfig                           |    3 +
 sound/soc/codecs/Makefile                          |    2 +
 sound/soc/codecs/cpcap.c                           | 1422 ++++++++++++++++++++
 4 files changed, 1446 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
 create mode 100644 sound/soc/codecs/cpcap.c

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..6b8cd616bd46
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
@@ -0,0 +1,19 @@
+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"
+  - vdd-supply : Phandle to audio regulator
+
+Example:
+
+&cpcap {
+	audio-codec {
+		compatible = "motorola,cpcap-audio-codec";
+		vdd-supply = <&vaudio>;
+	};
+};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 883ed4c8a551..e3d08ebeb37c 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -403,6 +403,9 @@ config SND_SOC_BT_SCO
 config SND_SOC_CQ0093VC
 	tristate
 
+config SND_SOC_CPCAP
+	tristate "Motorola CPCAP codec"
+
 config SND_SOC_CS35L32
 	tristate "Cirrus Logic CS35L32 CODEC"
 	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 28a63fdaf982..8750fd9c7dca 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -36,6 +36,7 @@ snd-soc-ak5386-objs := ak5386.o
 snd-soc-arizona-objs := arizona.o
 snd-soc-bt-sco-objs := bt-sco.o
 snd-soc-cq93vc-objs := cq93vc.o
+snd-soc-cpcap-objs := cpcap.o
 snd-soc-cs35l32-objs := cs35l32.o
 snd-soc-cs35l33-objs := cs35l33.o
 snd-soc-cs35l34-objs := cs35l34.o
@@ -271,6 +272,7 @@ 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..15ec732e0950
--- /dev/null
+++ b/sound/soc/codecs/cpcap.c
@@ -0,0 +1,1422 @@
+/*
+ * ALSA SoC CPCAP codec driver
+ *
+ * Copyright (C) 2017 Sebastian Reichel <sre@kernel.org>
+ *
+ * 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
+
+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_audio_read(struct cpcap_audio *cpcap, u16 reg, u16 *val)
+{
+	struct snd_soc_codec *codec = cpcap->codec;
+	unsigned int value;
+	int err;
+
+	err = regmap_read(cpcap->regmap, reg, &value);
+	if (err) {
+		dev_err(codec->dev, "read failed: reg=%04x err=%d", reg, err);
+		return err;
+	}
+
+	*val = value;
+	return 0;
+}
+
+static int cpcap_audio_write(struct cpcap_audio *cpcap,
+			     u16 reg, u16 mask, u16 val)
+{
+	struct snd_soc_codec *codec = cpcap->codec;
+	struct device *dev = codec->dev;
+	int err;
+
+	err = regmap_update_bits(cpcap->regmap, reg, mask, val);
+	if (err)
+		dev_err(dev, "write failed: reg=%04x mask=%04x val=%04x err=%d",
+			reg, mask, val, err);
+
+	return err;
+}
+
+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;
+
+	/* 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);
+		err += regmap_write(cpcap->regmap, CPCAP_REG_ST_TEST1,
+				    STM_STDAC_EN_ST_TEST1_PRE);
+		return err;
+	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);
+		err += regmap_write(cpcap->regmap, CPCAP_REG_TEST,
+				    STM_STDAC_EN_TEST_POST);
+		return err;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+/* Capture Gain Control: 0dB to 31dB in 1dB steps */
+static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0);
+
+/* Playback Gain Control: -33dB to 12dB in 3dB steps */
+static DECLARE_TLV_DB_SCALE(vol_tlv, -3300, 300, 0);
+
+static const char * const cpcap_phase_invert_texts[] = {
+	"Non Inverted", "Inverted"
+};
+static const SOC_ENUM_SINGLE_DECL(cpcap_hifi_l_phase_inv_enum,
+	CPCAP_REG_RXSDOA, CPCAP_BIT_MONO_DAC0, cpcap_phase_invert_texts);
+static const 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_onoff_texts[] = {
+	"Off", "On"
+};
+static const SOC_ENUM_SINGLE_DECL(cpcap_ext_cap_l_enum,
+	CPCAP_REG_TXI, CPCAP_BIT_RX_L_ENCODE, cpcap_onoff_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_ext_cap_r_enum,
+	CPCAP_REG_TXI, CPCAP_BIT_RX_R_ENCODE, cpcap_onoff_texts);
+static const struct snd_kcontrol_new cpcap_extr_cap_control =
+	SOC_DAPM_ENUM("Ext Right Capture", cpcap_ext_cap_r_enum);
+static const struct snd_kcontrol_new cpcap_extl_cap_control =
+	SOC_DAPM_ENUM("Ext Left Capture", cpcap_ext_cap_l_enum);
+
+static const char * const cpcap_mic2_texts[] = {
+	"Off", "Mic 2"
+};
+static const SOC_ENUM_SINGLE_DECL(cpcap_mic_l_enum,
+	CPCAP_REG_TXI, CPCAP_BIT_MIC2_MUX, cpcap_mic2_texts);
+static const struct snd_kcontrol_new cpcap_micl_control =
+	SOC_DAPM_ENUM("Mic Left source", cpcap_mic_l_enum);
+
+static const char * const cpcap_mic1_texts[] = {
+	"Off", "Mic 1", "Headset Mic", "(unused)", "EMU Mic"
+};
+static const SOC_ENUM_SINGLE_DECL(cpcap_mic_r_enum,
+	CPCAP_REG_TXI, CPCAP_BIT_MIC1_MUX, cpcap_mic1_texts);
+static const struct snd_kcontrol_new cpcap_micr_control =
+	SOC_DAPM_ENUM("Mic Right source", cpcap_mic_r_enum);
+
+static const char * const cpcap_output_mux_texts[] = {
+	"Off", "Voice", "HiFi", "Ext"
+};
+
+/*
+ * 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.
+ * Selecting enabling multiple inputs will result in no audio being forwarded.
+ */
+static const SOC_ENUM_SINGLE_DECL(cpcap_earpiece_mux_enum, 0, 0,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_spkr_r_mux_enum, CPCAP_REG_RXSDOA, 1,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_spkr_l_mux_enum, 0, 2,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_line_r_mux_enum, 0, 3,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_line_l_mux_enum, 0, 4,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_hs_r_mux_enum, 0, 5,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_hs_l_mux_enum, 0, 6,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_emu_l_mux_enum, 0, 7,
+				  cpcap_output_mux_texts);
+static const SOC_ENUM_SINGLE_DECL(cpcap_emu_r_mux_enum, 0, 8,
+				  cpcap_output_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;
+	u16 reg_voice, reg_hifi, reg_ext, status;
+	int err;
+
+	err = cpcap_audio_read(cpcap, CPCAP_REG_RXCOA, &reg_voice);
+	if (err)
+		return err;
+	cpcap_audio_read(cpcap, CPCAP_REG_RXSDOA, &reg_hifi);
+	if (err)
+		return err;
+	cpcap_audio_read(cpcap, CPCAP_REG_RXEPOA, &reg_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 snd_soc_card *card = dapm->card;
+	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 = cpcap_audio_write(cpcap, CPCAP_REG_RXCOA, mask, reg_voice);
+	if (err)
+		return err;
+	err = cpcap_audio_write(cpcap, CPCAP_REG_RXSDOA, mask, reg_hifi);
+	if (err)
+		return err;
+	err = cpcap_audio_write(cpcap, CPCAP_REG_RXEPOA, mask, reg_ext);
+	if (err)
+		return err;
+
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, muxval, e, NULL);
+	soc_dpcm_runtime_update(card);
+
+	return 0;
+}
+
+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("vdd", 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 MUX */
+	SND_SOC_DAPM_MUX("Right Capture Route",
+		SND_SOC_NOPM, 0, 0, &cpcap_micr_control),
+	SND_SOC_DAPM_MUX("Left Capture Route",
+		SND_SOC_NOPM, 0, 0, &cpcap_micl_control),
+	SND_SOC_DAPM_MUX("Ext Right Capture Route",
+		SND_SOC_NOPM, 0, 0, &cpcap_extr_cap_control),
+	SND_SOC_DAPM_MUX("Ext Left Capture Route",
+		SND_SOC_NOPM, 0, 0, &cpcap_extl_cap_control),
+
+	/* 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 Routees */
+	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, "vdd"},
+	{"Voice PGA", NULL, "vdd"},
+	{"Ext Right PGA", NULL, "vdd"},
+	{"Ext Left PGA", NULL, "vdd"},
+	{"Microphone 1 PGA", NULL, "vdd"},
+	{"Microphone 2 PGA", NULL, "vdd"},
+
+	/* 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"},
+	{"Left Capture Route", "Mic 2", "MICL"},
+
+	/* Ext -> Ext Route */
+	{"Ext Left Capture Route", "On", "EXTL"},
+	{"Ext Right Capture Route", "On", "EXTR"},
+
+	/* 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"},
+
+	/* Ext Route -> ADC */
+	{"ADC Right", NULL, "Ext Right Capture Route"},
+	{"ADC Left", NULL, "Ext Left Capture Route"},
+
+	/* 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 = cpcap_audio_write(cpcap, 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 = cpcap_audio_write(cpcap, 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 = cpcap_audio_write(cpcap, 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;
+
+	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 = cpcap_audio_write(cpcap, sampreg, sampmask, sampval | sampreset);
+	if (err)
+		return err;
+
+	/* Wait for clock tree reset to complete */
+	mdelay(CLOCK_TREE_RESET_TIME);
+
+	err = cpcap_audio_read(cpcap, sampreg, &sampval);
+	if (err)
+		return err;
+
+	if (sampval & sampreset) {
+		dev_err(codec->dev, "reset self-clear failed: %04x", sampval);
+		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 cpcap_audio_write(cpcap, 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 = mute ? 0 : BIT(CPCAP_BIT_ST_DAC_SW);
+
+	dev_dbg(codec->dev, "HiFi mute: %d", mute);
+	return cpcap_audio_write(cpcap, 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 = cpcap_audio_write(cpcap, 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 = cpcap_audio_write(cpcap, 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 = mute ? 0 : BIT(CPCAP_BIT_CDC_SW);
+
+	dev_dbg(codec->dev, "Voice mute: %d", mute);
+	return cpcap_audio_write(cpcap, 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_audio_reset(struct snd_soc_codec *codec,
+			     bool swap_dai_configuration)
+{
+	struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+	u16 cdi_def = 0x000;
+	u16 sdacdi_def = 0x0000;
+	int err = 0;
+
+	dev_dbg(codec->dev, "init audio codec");
+
+	if (!swap_dai_configuration) {
+		/* Codec on DAI0, HiFi on DAI1 */
+		cdi_def &= ~BIT(CPCAP_BIT_DIG_AUD_IN);
+		sdacdi_def |= BIT(CPCAP_BIT_DIG_AUD_IN_ST_DAC);
+	} else {
+		/* Codec on DAI1, HiFi on DAI0 */
+		cdi_def |= BIT(CPCAP_BIT_DIG_AUD_IN);
+		sdacdi_def &= ~BIT(CPCAP_BIT_DIG_AUD_IN_ST_DAC);
+	}
+
+	err += cpcap_audio_write(cpcap, CPCAP_REG_CC, 0xFFFF, 0x0000);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_CDI, 0xBFFF, cdi_def);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_SDAC, 0x0FFF, 0x0000);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_SDACDI, 0x3FFF, sdacdi_def);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_TXI, 0x0FDF, 0x0000);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_TXMP, 0x0FFF, 0x0400);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_RXOA, 0x01FF, 0x0000);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_RXVC, 0xFF3C, 0x0000);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_RXCOA, 0x07FF, 0x0000);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_RXSDOA, 0x1FFF, 0x0000);
+	err += cpcap_audio_write(cpcap, CPCAP_REG_RXEPOA, 0x7FFF, 0x0000);
+	/* skip CPCAP_REG_RXLL */
+	err += cpcap_audio_write(cpcap, CPCAP_REG_A2LA,
+		BIT(CPCAP_BIT_A2_FREE_RUN), BIT(CPCAP_BIT_A2_FREE_RUN));
+
+	if (err)
+		return err;
+
+	/* setup default settings */
+	err += cpcap_set_sysclk(cpcap, CPCAP_DAI_HIFI, 0, 26000000);
+	err += cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, 0, 26000000);
+	err += cpcap_set_samprate(cpcap, CPCAP_DAI_HIFI, 48000);
+	err += cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, 48000);
+
+	return err;
+}
+
+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");
-- 
2.13.2

Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Tony Lindgren]

* Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170707 09:43]:
> 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.

I needed the patch below for modular .config. Other than that,
plays music fine for me, so feel free to add:

Acked-by: Tony Lindgren <tony@atomide.com>

8< -----------------------
>From tony Mon Sep 17 00:00:00 2001
From: Tony Lindgren <tony@atomide.com>
Date: Sat, 8 Jul 2017 21:43:46 -0700
Subject: [PATCH] ALSA: pcm: Export soc_dpcm_runtime_update

Some drivers may need to use this from loadable modules. Otherwise we
will get:

ERROR: "soc_dpcm_runtime_update" [sound/soc/codecs/snd-soc-cpcap.ko]
undefined!

Signed-off-by: Tony Lindgren <tony@atomide.com>
---
 sound/soc/soc-pcm.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2539,6 +2539,8 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
 	mutex_unlock(&card->mutex);
 	return 0;
 }
+EXPORT_SYMBOL_GPL(soc_dpcm_runtime_update);
+
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 {
 	struct snd_soc_dpcm *dpcm;
-- 
2.13.2

Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Tony Lindgren]

* Sebastian Reichel <sebastian.reichel-ZGY8ohtN/8pPYcu2f3hruQ@public.gmane.org> [170707 09:43]:
> 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.

I needed the patch below for modular .config. Other than that,
plays music fine for me, so feel free to add:

Acked-by: Tony Lindgren <tony-4v6yS6AI5VpBDgjK7y7TUQ@public.gmane.org>

8< -----------------------
>From tony Mon Sep 17 00:00:00 2001
From: Tony Lindgren <tony-4v6yS6AI5VpBDgjK7y7TUQ@public.gmane.org>
Date: Sat, 8 Jul 2017 21:43:46 -0700
Subject: [PATCH] ALSA: pcm: Export soc_dpcm_runtime_update

Some drivers may need to use this from loadable modules. Otherwise we
will get:

ERROR: "soc_dpcm_runtime_update" [sound/soc/codecs/snd-soc-cpcap.ko]
undefined!

Signed-off-by: Tony Lindgren <tony-4v6yS6AI5VpBDgjK7y7TUQ@public.gmane.org>
---
 sound/soc/soc-pcm.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2539,6 +2539,8 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
 	mutex_unlock(&card->mutex);
 	return 0;
 }
+EXPORT_SYMBOL_GPL(soc_dpcm_runtime_update);
+
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 {
 	struct snd_soc_dpcm *dpcm;
-- 
2.13.2
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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Rob Herring]

On Fri, Jul 07, 2017 at 06:42:27PM +0200, Sebastian Reichel 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 <sebastian.reichel@collabora.co.uk>
> ---
>  .../bindings/sound/motorola,cpcap-audio-codec.txt  |   19 +

Acked-by: Rob Herring <robh@kernel.org>

>  sound/soc/codecs/Kconfig                           |    3 +
>  sound/soc/codecs/Makefile                          |    2 +
>  sound/soc/codecs/cpcap.c                           | 1422 ++++++++++++++++++++
>  4 files changed, 1446 insertions(+)
>  create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
>  create mode 100644 sound/soc/codecs/cpcap.c

Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Rob Herring]

On Fri, Jul 07, 2017 at 06:42:27PM +0200, Sebastian Reichel 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 <sebastian.reichel-ZGY8ohtN/8pPYcu2f3hruQ@public.gmane.org>
> ---
>  .../bindings/sound/motorola,cpcap-audio-codec.txt  |   19 +

Acked-by: Rob Herring <robh-DgEjT+Ai2ygdnm+yROfE0A@public.gmane.org>

>  sound/soc/codecs/Kconfig                           |    3 +
>  sound/soc/codecs/Makefile                          |    2 +
>  sound/soc/codecs/cpcap.c                           | 1422 ++++++++++++++++++++
>  4 files changed, 1446 insertions(+)
>  create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
>  create mode 100644 sound/soc/codecs/cpcap.c
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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 2645 bytes --]

On Fri, Jul 07, 2017 at 06:42:27PM +0200, Sebastian Reichel wrote:

>  snd-soc-cq93vc-objs := cq93vc.o
> +snd-soc-cpcap-objs := cpcap.o

Please keep Kconfig and Makefile lexically sorted.

> +static int cpcap_audio_write(struct cpcap_audio *cpcap,
> +			     u16 reg, u16 mask, u16 val)
> +{
> +	struct snd_soc_codec *codec = cpcap->codec;
> +	struct device *dev = codec->dev;
> +	int err;
> +
> +	err = regmap_update_bits(cpcap->regmap, reg, mask, val);
> +	if (err)
> +		dev_err(dev, "write failed: reg=%04x mask=%04x val=%04x err=%d",
> +			reg, mask, val, err);
> +
> +	return err;
> +}

If we're going to have wrappers for the regmap functions it'd be good if
they were named in a similar way to the functions that they wrap, just
for clarity.  They're also not used consistently (and TBH I'm not sure
what they buy but if you want them that's fine).

> +	switch (event) {
> +	case SND_SOC_DAPM_PRE_PMU:
> +		err += regmap_write(cpcap->regmap, CPCAP_REG_TEST,
> +				    STM_STDAC_EN_TEST_PRE);
> +		err += regmap_write(cpcap->regmap, CPCAP_REG_ST_TEST1,
> +				    STM_STDAC_EN_ST_TEST1_PRE);
> +		return err;

This'll return a nonsense error code if both error out, better to do
something like

	if (!err)
		err = regmap_write(...

> +static const char * const cpcap_onoff_texts[] = {
> +	"Off", "On"
> +};
> +static const SOC_ENUM_SINGLE_DECL(cpcap_ext_cap_l_enum,
> +	CPCAP_REG_TXI, CPCAP_BIT_RX_L_ENCODE, cpcap_onoff_texts);
> +static const SOC_ENUM_SINGLE_DECL(cpcap_ext_cap_r_enum,
> +	CPCAP_REG_TXI, CPCAP_BIT_RX_R_ENCODE, cpcap_onoff_texts);
> +static const struct snd_kcontrol_new cpcap_extr_cap_control =
> +	SOC_DAPM_ENUM("Ext Right Capture", cpcap_ext_cap_r_enum);
> +static const struct snd_kcontrol_new cpcap_extl_cap_control =
> +	SOC_DAPM_ENUM("Ext Left Capture", cpcap_ext_cap_l_enum);

Why are these enums and not simple Switch controls?  They appear to be
muxes with only one arm connected.

> +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 = mute ? 0 : BIT(CPCAP_BIT_ST_DAC_SW);

Please write a normal if statement, this was a bit confusing at first
glance.

> +#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

If this is part of a MFD shouldn't the parent device register it without
it needing to be in the DT?

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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 2645 bytes --]

On Fri, Jul 07, 2017 at 06:42:27PM +0200, Sebastian Reichel wrote:

>  snd-soc-cq93vc-objs := cq93vc.o
> +snd-soc-cpcap-objs := cpcap.o

Please keep Kconfig and Makefile lexically sorted.

> +static int cpcap_audio_write(struct cpcap_audio *cpcap,
> +			     u16 reg, u16 mask, u16 val)
> +{
> +	struct snd_soc_codec *codec = cpcap->codec;
> +	struct device *dev = codec->dev;
> +	int err;
> +
> +	err = regmap_update_bits(cpcap->regmap, reg, mask, val);
> +	if (err)
> +		dev_err(dev, "write failed: reg=%04x mask=%04x val=%04x err=%d",
> +			reg, mask, val, err);
> +
> +	return err;
> +}

If we're going to have wrappers for the regmap functions it'd be good if
they were named in a similar way to the functions that they wrap, just
for clarity.  They're also not used consistently (and TBH I'm not sure
what they buy but if you want them that's fine).

> +	switch (event) {
> +	case SND_SOC_DAPM_PRE_PMU:
> +		err += regmap_write(cpcap->regmap, CPCAP_REG_TEST,
> +				    STM_STDAC_EN_TEST_PRE);
> +		err += regmap_write(cpcap->regmap, CPCAP_REG_ST_TEST1,
> +				    STM_STDAC_EN_ST_TEST1_PRE);
> +		return err;

This'll return a nonsense error code if both error out, better to do
something like

	if (!err)
		err = regmap_write(...

> +static const char * const cpcap_onoff_texts[] = {
> +	"Off", "On"
> +};
> +static const SOC_ENUM_SINGLE_DECL(cpcap_ext_cap_l_enum,
> +	CPCAP_REG_TXI, CPCAP_BIT_RX_L_ENCODE, cpcap_onoff_texts);
> +static const SOC_ENUM_SINGLE_DECL(cpcap_ext_cap_r_enum,
> +	CPCAP_REG_TXI, CPCAP_BIT_RX_R_ENCODE, cpcap_onoff_texts);
> +static const struct snd_kcontrol_new cpcap_extr_cap_control =
> +	SOC_DAPM_ENUM("Ext Right Capture", cpcap_ext_cap_r_enum);
> +static const struct snd_kcontrol_new cpcap_extl_cap_control =
> +	SOC_DAPM_ENUM("Ext Left Capture", cpcap_ext_cap_l_enum);

Why are these enums and not simple Switch controls?  They appear to be
muxes with only one arm connected.

> +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 = mute ? 0 : BIT(CPCAP_BIT_ST_DAC_SW);

Please write a normal if statement, this was a bit confusing at first
glance.

> +#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

If this is part of a MFD shouldn't the parent device register it without
it needing to be in the DT?

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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Tony Lindgren]

* Mark Brown <broonie@kernel.org> [170710 10:52]:
> On Fri, Jul 07, 2017 at 06:42:27PM +0200, Sebastian Reichel wrote:
> > +#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
> 
> If this is part of a MFD shouldn't the parent device register it without
> it needing to be in the DT?

Having the MFD core part just do devm_of_platform_populate() leaves out
dependencies between the MFD core and it's child devices.

There are also a lot of board specific configuration to be done for many
child devices such as ADC wiring, GPIO pins used, and macro firmware
usage.

Not sure what all board specific stuff needs to be configured for this
driver yet, but it seems at least the macro interrupt wiring needs to
be configured. I would not be surprised to find GPIOs or ADC being used
for some connector detection too.

Regards,

Tony

Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Tony Lindgren]

* Mark Brown <broonie@kernel.org> [170710 10:52]:
> On Fri, Jul 07, 2017 at 06:42:27PM +0200, Sebastian Reichel wrote:
> > +#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
> 
> If this is part of a MFD shouldn't the parent device register it without
> it needing to be in the DT?

Having the MFD core part just do devm_of_platform_populate() leaves out
dependencies between the MFD core and it's child devices.

There are also a lot of board specific configuration to be done for many
child devices such as ADC wiring, GPIO pins used, and macro firmware
usage.

Not sure what all board specific stuff needs to be configured for this
driver yet, but it seems at least the macro interrupt wiring needs to
be configured. I would not be surprised to find GPIOs or ADC being used
for some connector detection too.

Regards,

Tony

Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 1184 bytes --]

On Mon, Jul 10, 2017 at 10:15:41PM -0700, Tony Lindgren wrote:
> * Mark Brown <broonie@kernel.org> [170710 10:52]:

> > If this is part of a MFD shouldn't the parent device register it without
> > it needing to be in the DT?

> Having the MFD core part just do devm_of_platform_populate() leaves out
> dependencies between the MFD core and it's child devices.

> There are also a lot of board specific configuration to be done for many
> child devices such as ADC wiring, GPIO pins used, and macro firmware
> usage.

> Not sure what all board specific stuff needs to be configured for this
> driver yet, but it seems at least the macro interrupt wiring needs to
> be configured. I would not be surprised to find GPIOs or ADC being used
> for some connector detection too.

We can use subnodes without compatible strings, that's not a problem,
but having to have compatible strings means we're encoding the way Linux
splits device drivers up into the DT which might not work for other OSs
or even future versions of Linux.  For example with CODEC drivers it's
common to have a good chunk of clock control in there which we might at
some point want to move over to the clock subsystem.

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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 1184 bytes --]

On Mon, Jul 10, 2017 at 10:15:41PM -0700, Tony Lindgren wrote:
> * Mark Brown <broonie@kernel.org> [170710 10:52]:

> > If this is part of a MFD shouldn't the parent device register it without
> > it needing to be in the DT?

> Having the MFD core part just do devm_of_platform_populate() leaves out
> dependencies between the MFD core and it's child devices.

> There are also a lot of board specific configuration to be done for many
> child devices such as ADC wiring, GPIO pins used, and macro firmware
> usage.

> Not sure what all board specific stuff needs to be configured for this
> driver yet, but it seems at least the macro interrupt wiring needs to
> be configured. I would not be surprised to find GPIOs or ADC being used
> for some connector detection too.

We can use subnodes without compatible strings, that's not a problem,
but having to have compatible strings means we're encoding the way Linux
splits device drivers up into the DT which might not work for other OSs
or even future versions of Linux.  For example with CODEC drivers it's
common to have a good chunk of clock control in there which we might at
some point want to move over to the clock subsystem.

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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Sebastian Reichel]

[-- Attachment #1: Type: text/plain, Size: 1651 bytes --]

Hi,

On Tue, Jul 11, 2017 at 11:49:11AM +0100, Mark Brown wrote:
> On Mon, Jul 10, 2017 at 10:15:41PM -0700, Tony Lindgren wrote:
> > * Mark Brown <broonie@kernel.org> [170710 10:52]:
> 
> > > If this is part of a MFD shouldn't the parent device register it without
> > > it needing to be in the DT?
> 
> > Having the MFD core part just do devm_of_platform_populate() leaves out
> > dependencies between the MFD core and it's child devices.
> 
> > There are also a lot of board specific configuration to be done for many
> > child devices such as ADC wiring, GPIO pins used, and macro firmware
> > usage.
> 
> > Not sure what all board specific stuff needs to be configured for this
> > driver yet, but it seems at least the macro interrupt wiring needs to
> > be configured. I would not be surprised to find GPIOs or ADC being used
> > for some connector detection too.
> 
> We can use subnodes without compatible strings, that's not a problem,
> but having to have compatible strings means we're encoding the way Linux
> splits device drivers up into the DT which might not work for other OSs
> or even future versions of Linux.  For example with CODEC drivers it's
> common to have a good chunk of clock control in there which we might at
> some point want to move over to the clock subsystem.

How is having a subnode without a compatible property different?
Sure we will bind the driver manually and make our code (a little
bit) more complex, but we still encoded the way Linux splits the
device into DT. My understanding is, that adding a node without a
compatible value is almost always not ok.

-- Sebastian

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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Sebastian Reichel]

[-- Attachment #1.1: Type: text/plain, Size: 1651 bytes --]

Hi,

On Tue, Jul 11, 2017 at 11:49:11AM +0100, Mark Brown wrote:
> On Mon, Jul 10, 2017 at 10:15:41PM -0700, Tony Lindgren wrote:
> > * Mark Brown <broonie@kernel.org> [170710 10:52]:
> 
> > > If this is part of a MFD shouldn't the parent device register it without
> > > it needing to be in the DT?
> 
> > Having the MFD core part just do devm_of_platform_populate() leaves out
> > dependencies between the MFD core and it's child devices.
> 
> > There are also a lot of board specific configuration to be done for many
> > child devices such as ADC wiring, GPIO pins used, and macro firmware
> > usage.
> 
> > Not sure what all board specific stuff needs to be configured for this
> > driver yet, but it seems at least the macro interrupt wiring needs to
> > be configured. I would not be surprised to find GPIOs or ADC being used
> > for some connector detection too.
> 
> We can use subnodes without compatible strings, that's not a problem,
> but having to have compatible strings means we're encoding the way Linux
> splits device drivers up into the DT which might not work for other OSs
> or even future versions of Linux.  For example with CODEC drivers it's
> common to have a good chunk of clock control in there which we might at
> some point want to move over to the clock subsystem.

How is having a subnode without a compatible property different?
Sure we will bind the driver manually and make our code (a little
bit) more complex, but we still encoded the way Linux splits the
device into DT. My understanding is, that adding a node without a
compatible value is almost always not ok.

-- Sebastian

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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Mark Brown]

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On Tue, Jul 11, 2017 at 02:13:31PM +0200, Sebastian Reichel wrote:

> How is having a subnode without a compatible property different?

You don't *need* to have the subnode, I was only mentioning that if for
some reason it was super useful for organizing the properties.

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Re: [PATCH 1/3] ASoC: codec: cpcap: new codec

[Mark Brown]

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On Tue, Jul 11, 2017 at 02:13:31PM +0200, Sebastian Reichel wrote:

> How is having a subnode without a compatible property different?

You don't *need* to have the subnode, I was only mentioning that if for
some reason it was super useful for organizing the properties.

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[PATCH 2/3] ARM: dts: motorola-cpcap-mapphone: add audio-codec

[Sebastian Reichel]

Add node for audio-codec to its DT file.

Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
 arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
index 1eb5da1dc8f0..86033aee8a29 100644
--- a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
+++ b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
@@ -68,6 +68,12 @@
 			};
 		};
 
+		cpcap_audio: audio-codec {
+			#sound-dai-cells = <1>;
+			compatible = "motorola,cpcap-audio-codec";
+			vdd-supply = <&vaudio>;
+		};
+
 		cpcap_rtc: rtc {
 			compatible = "motorola,cpcap-rtc";
 
-- 
2.13.2

[PATCH 2/3] ARM: dts: motorola-cpcap-mapphone: add audio-codec

[Sebastian Reichel]

Add node for audio-codec to its DT file.

Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
 arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
index 1eb5da1dc8f0..86033aee8a29 100644
--- a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
+++ b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
@@ -68,6 +68,12 @@
 			};
 		};
 
+		cpcap_audio: audio-codec {
+			#sound-dai-cells = <1>;
+			compatible = "motorola,cpcap-audio-codec";
+			vdd-supply = <&vaudio>;
+		};
+
 		cpcap_rtc: rtc {
 			compatible = "motorola,cpcap-rtc";
 
-- 
2.13.2

[PATCH 3/3] ARM: dts: omap4-droid4: add soundcard

[Sebastian Reichel]

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 <sebastian.reichel@collabora.co.uk>
---
 arch/arm/boot/dts/omap4-droid4-xt894.dts | 76 ++++++++++++++++++++++++++++++++
 1 file changed, 76 insertions(+)

diff --git a/arch/arm/boot/dts/omap4-droid4-xt894.dts b/arch/arm/boot/dts/omap4-droid4-xt894.dts
index 10ca1c174995..642b7ed1713b 100644
--- a/arch/arm/boot/dts/omap4-droid4-xt894.dts
+++ b/arch/arm/boot/dts/omap4-droid4-xt894.dts
@@ -116,6 +116,50 @@
 
 		};
 	};
+
+	soundcard {
+		compatible = "simple-audio-card";
+		simple-audio-card,name = "Droid 4 Audio";
+
+		simple-audio-card,widgets =
+			"Speaker", "Earpiece",
+			"Speaker", "Loudspeaker",
+			"Headphone", "Headphone Jack",
+			"Microphone", "Internal Mic";
+
+		simple-audio-card,routing =
+			"Earpiece", "EP",
+			"Loudspeaker", "SPKR",
+			"Headphone Jack", "HSL",
+			"Headphone Jack", "HSR",
+			"MICR", "Internal Mic";
+
+		simple-audio-card,dai-link@0 {
+			format = "i2s";
+			frame-master = <&codec0>;
+			bitclock-master = <&codec0>;
+			cpu {
+				sound-dai = <&mcbsp2>;
+			};
+			codec0: codec {
+				sound-dai = <&cpcap_audio 0>;
+			};
+		};
+
+		simple-audio-card,dai-link@1 {
+			format = "dsp_a";
+			frame-master = <&codec1>;
+			bitclock-master = <&codec1>;
+			bitclock-inversion;
+
+			cpu {
+				sound-dai = <&mcbsp3>;
+			};
+			codec1: codec {
+				sound-dai = <&cpcap_audio 1>;
+			};
+		};
+	};
 };
 
 &dss {
@@ -479,6 +523,24 @@
 		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 {
@@ -552,3 +614,17 @@
 				  "0", "0", "1";
 	};
 };
+
+&mcbsp2 {
+	#sound-dai-cells = <0>;
+	pinctrl-names = "default";
+	pinctrl-0 = <&mcbsp2_pins>;
+	status = "okay";
+};
+
+&mcbsp3 {
+	#sound-dai-cells = <0>;
+	pinctrl-names = "default";
+	pinctrl-0 = <&mcbsp3_pins>;
+	status = "okay";
+};
-- 
2.13.2

Re: [PATCH 3/3] ARM: dts: omap4-droid4: add soundcard

[Mark Brown]

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On Fri, Jul 07, 2017 at 06:42:29PM +0200, Sebastian Reichel wrote:

> +	soundcard {
> +		compatible = "simple-audio-card";
> +		simple-audio-card,name = "Droid 4 Audio";

New systems should be using the of-graph card which is more flexible and
extensible than simple-card.

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Re: [PATCH 3/3] ARM: dts: omap4-droid4: add soundcard

[Pavel Machek]

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On Mon 2017-07-10 13:26:02, Mark Brown wrote:
> On Fri, Jul 07, 2017 at 06:42:29PM +0200, Sebastian Reichel wrote:
> 
> > +	soundcard {
> > +		compatible = "simple-audio-card";
> > +		simple-audio-card,name = "Droid 4 Audio";
> 
> New systems should be using the of-graph card which is more flexible and
> extensible than simple-card.

Well, if simple-card does what is needed here, and dts looks
reasonable, is there reason for anything more complex?

Anyway, congratulations for getting the sound to work!

									Pavel


-- 
(english) http://www.livejournal.com/~pavelmachek
(cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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Re: [PATCH 3/3] ARM: dts: omap4-droid4: add soundcard

[Pavel Machek]

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On Mon 2017-07-10 13:26:02, Mark Brown wrote:
> On Fri, Jul 07, 2017 at 06:42:29PM +0200, Sebastian Reichel wrote:
> 
> > +	soundcard {
> > +		compatible = "simple-audio-card";
> > +		simple-audio-card,name = "Droid 4 Audio";
> 
> New systems should be using the of-graph card which is more flexible and
> extensible than simple-card.

Well, if simple-card does what is needed here, and dts looks
reasonable, is there reason for anything more complex?

Anyway, congratulations for getting the sound to work!

									Pavel


-- 
(english) http://www.livejournal.com/~pavelmachek
(cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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Re: [PATCH 3/3] ARM: dts: omap4-droid4: add soundcard

[Sebastian Reichel]

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Hi,

On Mon, Jul 24, 2017 at 09:19:54PM +0200, Pavel Machek wrote:
> On Mon 2017-07-10 13:26:02, Mark Brown wrote:
> > On Fri, Jul 07, 2017 at 06:42:29PM +0200, Sebastian Reichel wrote:
> > 
> > > +	soundcard {
> > > +		compatible = "simple-audio-card";
> > > +		simple-audio-card,name = "Droid 4 Audio";
> > 
> > New systems should be using the of-graph card which is more flexible and
> > extensible than simple-card.
> 
> Well, if simple-card does what is needed here, and dts looks
> reasonable, is there reason for anything more complex?

There is already a PATCHv2 for this patchset, that switched to
the graph based card.

> Anyway, congratulations for getting the sound to work!

Thanks.

-- Sebastian

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Re: [PATCH 3/3] ARM: dts: omap4-droid4: add soundcard

[Sebastian Reichel]

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Hi,

On Mon, Jul 24, 2017 at 09:19:54PM +0200, Pavel Machek wrote:
> On Mon 2017-07-10 13:26:02, Mark Brown wrote:
> > On Fri, Jul 07, 2017 at 06:42:29PM +0200, Sebastian Reichel wrote:
> > 
> > > +	soundcard {
> > > +		compatible = "simple-audio-card";
> > > +		simple-audio-card,name = "Droid 4 Audio";
> > 
> > New systems should be using the of-graph card which is more flexible and
> > extensible than simple-card.
> 
> Well, if simple-card does what is needed here, and dts looks
> reasonable, is there reason for anything more complex?

There is already a PATCHv2 for this patchset, that switched to
the graph based card.

> Anyway, congratulations for getting the sound to work!

Thanks.

-- Sebastian

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Re: [PATCH 0/3] Motorola Droid 4 Audio Support

[Tony Lindgren]

* Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170707 09:43]:
> Hi,
> 
> I got working sound on Droid 4 with mainline \o/. The codec is
> currently missing support for detecting if something has been
> plugged into the 3.5mm connector, since that seems to require
> some closed source firmware and needs further investigation. I
> think this can be added later.

Hey that's great! I'll give it a try this weekend.. Does that mean
that 3G voice calls work too now or is something more needed there?

For the CPCAP PMIC macro interrupts I think it's best to set up
a separate driver as it seems separate from the core CPCAP
functionality. So I think we can just move the unused "cpcap-m2"
IRQ banks out of motorola-cpcap.c and put them into a separate child
driver that loads it's firmware on init and provides interrupts for
the 3.5mm connector.

Regards,

Tony

Re: [PATCH 0/3] Motorola Droid 4 Audio Support

[Tony Lindgren]

* Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170707 09:43]:
> Hi,
> 
> I got working sound on Droid 4 with mainline \o/. The codec is
> currently missing support for detecting if something has been
> plugged into the 3.5mm connector, since that seems to require
> some closed source firmware and needs further investigation. I
> think this can be added later.

Hey that's great! I'll give it a try this weekend.. Does that mean
that 3G voice calls work too now or is something more needed there?

For the CPCAP PMIC macro interrupts I think it's best to set up
a separate driver as it seems separate from the core CPCAP
functionality. So I think we can just move the unused "cpcap-m2"
IRQ banks out of motorola-cpcap.c and put them into a separate child
driver that loads it's firmware on init and provides interrupts for
the 3.5mm connector.

Regards,

Tony

Re: [PATCH 0/3] Motorola Droid 4 Audio Support

[Sebastian Reichel]

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Hi,

On Fri, Jul 07, 2017 at 10:27:53PM -0700, Tony Lindgren wrote:
> * Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170707 09:43]:
> > I got working sound on Droid 4 with mainline \o/. The codec is
> > currently missing support for detecting if something has been
> > plugged into the 3.5mm connector, since that seems to require
> > some closed source firmware and needs further investigation. I
> > think this can be added later.
> 
> Hey that's great! I'll give it a try this weekend.. Does that mean
> that 3G voice calls work too now or is something more needed there?

I added some text about that in the 3rd patch. Basically the codec driver
is ready (maybe a few quirks will be needed, though), but the soundcard
driver is not. It currently assumes CPU <-> Voice Codec, but
actually its a network of CPU, Voice Codec, Modem(s?) and Bluetooth.

> For the CPCAP PMIC macro interrupts I think it's best to set up
> a separate driver as it seems separate from the core CPCAP
> functionality. So I think we can just move the unused "cpcap-m2"
> IRQ banks out of motorola-cpcap.c and put them into a separate child
> driver that loads it's firmware on init and provides interrupts for
> the 3.5mm connector.

I also think the macro block fw handling should not be handled in
the driver for the audio codec. IIRC it is also used to provide a
blink feature for the status LEDs. OTOH some of the registers of the
audio-codec are also used for 3.5mm detection. I also did not yet look
into the details of this one.

-- Sebastian

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Re: [PATCH 0/3] Motorola Droid 4 Audio Support

[Sebastian Reichel]

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Hi,

On Fri, Jul 07, 2017 at 10:27:53PM -0700, Tony Lindgren wrote:
> * Sebastian Reichel <sebastian.reichel-ZGY8ohtN/8pPYcu2f3hruQ@public.gmane.org> [170707 09:43]:
> > I got working sound on Droid 4 with mainline \o/. The codec is
> > currently missing support for detecting if something has been
> > plugged into the 3.5mm connector, since that seems to require
> > some closed source firmware and needs further investigation. I
> > think this can be added later.
> 
> Hey that's great! I'll give it a try this weekend.. Does that mean
> that 3G voice calls work too now or is something more needed there?

I added some text about that in the 3rd patch. Basically the codec driver
is ready (maybe a few quirks will be needed, though), but the soundcard
driver is not. It currently assumes CPU <-> Voice Codec, but
actually its a network of CPU, Voice Codec, Modem(s?) and Bluetooth.

> For the CPCAP PMIC macro interrupts I think it's best to set up
> a separate driver as it seems separate from the core CPCAP
> functionality. So I think we can just move the unused "cpcap-m2"
> IRQ banks out of motorola-cpcap.c and put them into a separate child
> driver that loads it's firmware on init and provides interrupts for
> the 3.5mm connector.

I also think the macro block fw handling should not be handled in
the driver for the audio codec. IIRC it is also used to provide a
blink feature for the status LEDs. OTOH some of the registers of the
audio-codec are also used for 3.5mm detection. I also did not yet look
into the details of this one.

-- Sebastian

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Re: [PATCH 0/3] Motorola Droid 4 Audio Support

[Tony Lindgren]

* Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170708 04:34]:
> Hi,
> 
> On Fri, Jul 07, 2017 at 10:27:53PM -0700, Tony Lindgren wrote:
> > * Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170707 09:43]:
> > > I got working sound on Droid 4 with mainline \o/. The codec is
> > > currently missing support for detecting if something has been
> > > plugged into the 3.5mm connector, since that seems to require
> > > some closed source firmware and needs further investigation. I
> > > think this can be added later.
> > 
> > Hey that's great! I'll give it a try this weekend.. Does that mean
> > that 3G voice calls work too now or is something more needed there?
> 
> I added some text about that in the 3rd patch. Basically the codec driver
> is ready (maybe a few quirks will be needed, though), but the soundcard
> driver is not. It currently assumes CPU <-> Voice Codec, but
> actually its a network of CPU, Voice Codec, Modem(s?) and Bluetooth.

OK

> > For the CPCAP PMIC macro interrupts I think it's best to set up
> > a separate driver as it seems separate from the core CPCAP
> > functionality. So I think we can just move the unused "cpcap-m2"
> > IRQ banks out of motorola-cpcap.c and put them into a separate child
> > driver that loads it's firmware on init and provides interrupts for
> > the 3.5mm connector.
> 
> I also think the macro block fw handling should not be handled in
> the driver for the audio codec. IIRC it is also used to provide a
> blink feature for the status LEDs. OTOH some of the registers of the
> audio-codec are also used for 3.5mm detection. I also did not yet look
> into the details of this one.

Also the battery driver has three mystery interrupts coming from the
macro. At least the cycle count is missing so maybe that too comes
from the macro.

Regards,

Tony

Re: [PATCH 0/3] Motorola Droid 4 Audio Support

[Tony Lindgren]

* Sebastian Reichel <sebastian.reichel-ZGY8ohtN/8pPYcu2f3hruQ@public.gmane.org> [170708 04:34]:
> Hi,
> 
> On Fri, Jul 07, 2017 at 10:27:53PM -0700, Tony Lindgren wrote:
> > * Sebastian Reichel <sebastian.reichel-ZGY8ohtN/8pPYcu2f3hruQ@public.gmane.org> [170707 09:43]:
> > > I got working sound on Droid 4 with mainline \o/. The codec is
> > > currently missing support for detecting if something has been
> > > plugged into the 3.5mm connector, since that seems to require
> > > some closed source firmware and needs further investigation. I
> > > think this can be added later.
> > 
> > Hey that's great! I'll give it a try this weekend.. Does that mean
> > that 3G voice calls work too now or is something more needed there?
> 
> I added some text about that in the 3rd patch. Basically the codec driver
> is ready (maybe a few quirks will be needed, though), but the soundcard
> driver is not. It currently assumes CPU <-> Voice Codec, but
> actually its a network of CPU, Voice Codec, Modem(s?) and Bluetooth.

OK

> > For the CPCAP PMIC macro interrupts I think it's best to set up
> > a separate driver as it seems separate from the core CPCAP
> > functionality. So I think we can just move the unused "cpcap-m2"
> > IRQ banks out of motorola-cpcap.c and put them into a separate child
> > driver that loads it's firmware on init and provides interrupts for
> > the 3.5mm connector.
> 
> I also think the macro block fw handling should not be handled in
> the driver for the audio codec. IIRC it is also used to provide a
> blink feature for the status LEDs. OTOH some of the registers of the
> audio-codec are also used for 3.5mm detection. I also did not yet look
> into the details of this one.

Also the battery driver has three mystery interrupts coming from the
macro. At least the cycle count is missing so maybe that too comes
from the macro.

Regards,

Tony

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