[PATCH v4 0/4] ASoC: MAX9860: new driver

[PATCH v4 0/4] ASoC: MAX9860: new driver

[Peter Rosin]

Hi!

I kept the sidetone changes in separate patches since I don't know
if I solved the DAPM-off issue cleanly enough. Also, a new driver may
still have a chance to make 4.7, but maybe it's too late for the DAPM
changes?

Finally, I wonder if the device tree crowd prefers seeing the whole of
a series such as this, or only the bindings patch?

Cheers,
Peter

New in v4:
- Split out the device tree bindings in its own patch.
- Move format check from ->startup to ->set_fmt.
- Don't spam the log on probe defer.
- Follow-up patches for the backwards sidetone control.

New in v3:
- The updated bindings file went missing in v2. Sorry for the confusion.

New in v2:
- Add comment about fall through when Integer Clock Mode is not possible.
- Drop export of max9860_probe.
- Ignore clk docs and read the mclk rate w/o enabling the clock.
- Manage the DVDDIO supply.

Peter Rosin (4):
  dt-bindings: sound: add bindings for the max9860 codec
  ASoC: MAX9860: new driver
  ASoC: dapm: support mixer controls with mute at non-zero value
  ASoC: MAX9860: add sidetone mixer control

 .../devicetree/bindings/sound/max9860.txt          |  28 +
 MAINTAINERS                                        |   7 +
 sound/soc/codecs/Kconfig                           |   6 +
 sound/soc/codecs/Makefile                          |   2 +
 sound/soc/codecs/max9860.c                         | 765 +++++++++++++++++++++
 sound/soc/codecs/max9860.h                         | 162 +++++
 sound/soc/soc-dapm.c                               |  38 +-
 7 files changed, 1005 insertions(+), 3 deletions(-)
 create mode 100644 Documentation/devicetree/bindings/sound/max9860.txt
 create mode 100644 sound/soc/codecs/max9860.c
 create mode 100644 sound/soc/codecs/max9860.h

-- 
2.1.4

[PATCH v4 1/4] dt-bindings: sound: add bindings for the max9860 codec

[Peter Rosin]

This adds the device tree binding documentation for the Maxim Integrated
MAX9860 mono audio voice codec.

Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Peter Rosin <peda@axentia.se>
---
 .../devicetree/bindings/sound/max9860.txt          | 28 ++++++++++++++++++++++
 1 file changed, 28 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/max9860.txt

diff --git a/Documentation/devicetree/bindings/sound/max9860.txt b/Documentation/devicetree/bindings/sound/max9860.txt
new file mode 100644
index 000000000000..e0d4e95e31b3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/max9860.txt
@@ -0,0 +1,28 @@
+MAX9860 Mono Audio Voice Codec
+
+Required properties:
+
+  - compatible : "maxim,max9860"
+
+  - reg : the I2C address of the device
+
+  - AVDD-supply, DVDD-supply and DVDDIO-supply : power supplies for
+    the device, as covered in bindings/regulator/regulator.txt
+
+  - clock-names : Required element: "mclk".
+
+  - clocks : A clock specifier for the clock connected as MCLK.
+
+Examples:
+
+	max9860: max9860@10 {
+		compatible = "maxim,max9860";
+		reg = <0x10>;
+
+		AVDD-supply = <&reg_1v8>;
+		DVDD-supply = <&reg_1v8>;
+		DVDDIO-supply = <&reg_3v0>;
+
+		clock-names = "mclk";
+		clocks = <&pck2>;
+	};
-- 
2.1.4

[PATCH v4 2/4] ASoC: MAX9860: new driver

[Peter Rosin]

This is a driver for the MAX9860 Mono Audio Voice Codec.

https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf

This driver does not support sidetone since the DVST register field is
backwards with the mute near the maximum level instead of the minimum.

Signed-off-by: Peter Rosin <peda@axentia.se>
---
 MAINTAINERS                |   7 +
 sound/soc/codecs/Kconfig   |   6 +
 sound/soc/codecs/Makefile  |   2 +
 sound/soc/codecs/max9860.c | 753 +++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/max9860.h | 162 ++++++++++
 5 files changed, 930 insertions(+)
 create mode 100644 sound/soc/codecs/max9860.c
 create mode 100644 sound/soc/codecs/max9860.h

diff --git a/MAINTAINERS b/MAINTAINERS
index a727d9959ecd..e433bc54eba2 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -7000,6 +7000,13 @@ F:	Documentation/devicetree/bindings/i2c/max6697.txt
 F:	drivers/hwmon/max6697.c
 F:	include/linux/platform_data/max6697.h
 
+MAX9860 MONO AUDIO VOICE CODEC DRIVER
+M:	Peter Rosin <peda@axentia.se>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/max9860.txt
+F:	sound/soc/codecs/max9860.*
+
 MAXIM MUIC CHARGER DRIVERS FOR EXYNOS BASED BOARDS
 M:	Krzysztof Kozlowski <k.kozlowski@samsung.com>
 L:	linux-pm@vger.kernel.org
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7ef3a0c16478..241a23c9aa9f 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX98925 if I2C
 	select SND_SOC_MAX98926 if I2C
 	select SND_SOC_MAX9850 if I2C
+	select SND_SOC_MAX9860 if I2C
 	select SND_SOC_MAX9768 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_MC13783 if MFD_MC13XXX
@@ -534,6 +535,11 @@ config SND_SOC_MAX98926
 config SND_SOC_MAX9850
 	tristate
 
+config SND_SOC_MAX9860
+	tristate "Maxim MAX9860 Mono Audio Voice Codec"
+	depends on I2C
+	select REGMAP_I2C
+
 config SND_SOC_PCM1681
 	tristate "Texas Instruments PCM1681 CODEC"
 	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 185a712a7fe7..e435f4df1787 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -78,6 +78,7 @@ snd-soc-max9867-objs := max9867.o
 snd-soc-max98925-objs := max98925.o
 snd-soc-max98926-objs := max98926.o
 snd-soc-max9850-objs := max9850.o
+snd-soc-max9860-objs := max9860.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
 snd-soc-nau8825-objs := nau8825.o
@@ -287,6 +288,7 @@ obj-$(CONFIG_SND_SOC_MAX9867)	+= snd-soc-max9867.o
 obj-$(CONFIG_SND_SOC_MAX98925)	+= snd-soc-max98925.o
 obj-$(CONFIG_SND_SOC_MAX98926)	+= snd-soc-max98926.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
+obj-$(CONFIG_SND_SOC_MAX9860)	+= snd-soc-max9860.o
 obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
 obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c
new file mode 100644
index 000000000000..2b0dd6a18dad
--- /dev/null
+++ b/sound/soc/codecs/max9860.c
@@ -0,0 +1,753 @@
+/*
+ * Driver for the MAX9860 Mono Audio Voice Codec
+ *
+ * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf
+ *
+ * The driver does not support sidetone since the DVST register field is
+ * backwards with the mute near the maximum level instead of the minimum.
+ *
+ * Author: Peter Rosin <peda@axentia.s>
+ *         Copyright 2016 Axentia Technologies
+ *
+ * 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/init.h>
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/i2c.h>
+#include <linux/regulator/consumer.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#include "max9860.h"
+
+struct max9860_priv {
+	struct regmap *regmap;
+	struct regulator *dvddio;
+	struct notifier_block dvddio_nb;
+	u8 psclk;
+	unsigned long pclk_rate;
+	int fmt;
+};
+
+static int max9860_dvddio_event(struct notifier_block *nb,
+				unsigned long event, void *data)
+{
+	struct max9860_priv *max9860 = container_of(nb, struct max9860_priv,
+						    dvddio_nb);
+	if (event & REGULATOR_EVENT_DISABLE) {
+		regcache_mark_dirty(max9860->regmap);
+		regcache_cache_only(max9860->regmap, true);
+	}
+
+	return 0;
+}
+
+static const struct reg_default max9860_reg_defaults[] = {
+	{ MAX9860_PWRMAN,       0x00 },
+	{ MAX9860_INTEN,        0x00 },
+	{ MAX9860_SYSCLK,       0x00 },
+	{ MAX9860_AUDIOCLKHIGH, 0x00 },
+	{ MAX9860_AUDIOCLKLOW,  0x00 },
+	{ MAX9860_IFC1A,        0x00 },
+	{ MAX9860_IFC1B,        0x00 },
+	{ MAX9860_VOICEFLTR,    0x00 },
+	{ MAX9860_DACATTN,      0x00 },
+	{ MAX9860_ADCLEVEL,     0x00 },
+	{ MAX9860_DACGAIN,      0x00 },
+	{ MAX9860_MICGAIN,      0x00 },
+	{ MAX9860_MICADC,       0x00 },
+	{ MAX9860_NOISEGATE,    0x00 },
+};
+
+static bool max9860_readable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX9860_INTRSTATUS ... MAX9860_MICGAIN:
+	case MAX9860_MICADC ... MAX9860_PWRMAN:
+	case MAX9860_REVISION:
+		return true;
+	}
+
+	return false;
+}
+
+static bool max9860_writeable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX9860_INTEN ... MAX9860_MICGAIN:
+	case MAX9860_MICADC ... MAX9860_PWRMAN:
+		return true;
+	}
+
+	return false;
+}
+
+static bool max9860_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX9860_INTRSTATUS:
+	case MAX9860_MICREADBACK:
+		return true;
+	}
+
+	return false;
+}
+
+static bool max9860_precious(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX9860_INTRSTATUS:
+		return true;
+	}
+
+	return false;
+}
+
+const struct regmap_config max9860_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.readable_reg = max9860_readable,
+	.writeable_reg = max9860_writeable,
+	.volatile_reg = max9860_volatile,
+	.precious_reg = max9860_precious,
+
+	.max_register = MAX9860_MAX_REGISTER,
+	.reg_defaults = max9860_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(max9860_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_RANGE(pam_tlv,
+	0, MAX9860_PAM_MAX - 1,             TLV_DB_SCALE_ITEM(-2000, 2000, 1),
+	MAX9860_PAM_MAX, MAX9860_PAM_MAX,   TLV_DB_SCALE_ITEM(3000, 0, 0));
+static const DECLARE_TLV_DB_SCALE(pgam_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(anth_tlv, -7600, 400, 1);
+static const DECLARE_TLV_DB_SCALE(agcth_tlv, -1800, 100, 0);
+
+static const char * const agchld_text[] = {
+	"AGC Disabled", "50ms", "100ms", "400ms"
+};
+
+static SOC_ENUM_SINGLE_DECL(agchld_enum, MAX9860_MICADC,
+			    MAX9860_AGCHLD_SHIFT, agchld_text);
+
+static const char * const agcsrc_text[] = {
+	"Left ADC", "Left/Right ADC"
+};
+
+static SOC_ENUM_SINGLE_DECL(agcsrc_enum, MAX9860_MICADC,
+			    MAX9860_AGCSRC_SHIFT, agcsrc_text);
+
+static const char * const agcatk_text[] = {
+	"3ms", "12ms", "50ms", "200ms"
+};
+
+static SOC_ENUM_SINGLE_DECL(agcatk_enum, MAX9860_MICADC,
+			    MAX9860_AGCATK_SHIFT, agcatk_text);
+
+static const char * const agcrls_text[] = {
+	"78ms", "156ms", "312ms", "625ms",
+	"1.25s", "2.5s", "5s", "10s"
+};
+
+static SOC_ENUM_SINGLE_DECL(agcrls_enum, MAX9860_MICADC,
+			    MAX9860_AGCRLS_SHIFT, agcrls_text);
+
+static const char * const filter_text[] = {
+	"Disabled",
+	"Elliptical HP 217Hz notch (16kHz)",
+	"Butterworth HP 500Hz (16kHz)",
+	"Elliptical HP 217Hz notch (8kHz)",
+	"Butterworth HP 500Hz (8kHz)",
+	"Butterworth HP 200Hz (48kHz)"
+};
+
+static SOC_ENUM_SINGLE_DECL(avflt_enum, MAX9860_VOICEFLTR,
+			    MAX9860_AVFLT_SHIFT, filter_text);
+
+static SOC_ENUM_SINGLE_DECL(dvflt_enum, MAX9860_VOICEFLTR,
+			    MAX9860_DVFLT_SHIFT, filter_text);
+
+static const struct snd_kcontrol_new max9860_controls[] = {
+SOC_SINGLE_TLV("Master Playback Volume", MAX9860_DACATTN,
+	       MAX9860_DVA_SHIFT, MAX9860_DVA_MUTE, 1, dva_tlv),
+SOC_SINGLE_TLV("DAC Gain Volume", MAX9860_DACGAIN,
+	       MAX9860_DVG_SHIFT, MAX9860_DVG_MAX, 0, dvg_tlv),
+SOC_DOUBLE_TLV("Line Capture Volume", MAX9860_ADCLEVEL,
+	       MAX9860_ADCLL_SHIFT, MAX9860_ADCRL_SHIFT, MAX9860_ADCxL_MIN, 1,
+	       adc_tlv),
+
+SOC_ENUM("AGC Hold Time", agchld_enum),
+SOC_ENUM("AGC/Noise Gate Source", agcsrc_enum),
+SOC_ENUM("AGC Attack Time", agcatk_enum),
+SOC_ENUM("AGC Release Time", agcrls_enum),
+
+SOC_SINGLE_TLV("Noise Gate Threshold Volume", MAX9860_NOISEGATE,
+	       MAX9860_ANTH_SHIFT, MAX9860_ANTH_MAX, 0, anth_tlv),
+SOC_SINGLE_TLV("AGC Signal Threshold Volume", MAX9860_NOISEGATE,
+	       MAX9860_AGCTH_SHIFT, MAX9860_AGCTH_MIN, 1, agcth_tlv),
+
+SOC_SINGLE_TLV("Mic PGA Volume", MAX9860_MICGAIN,
+	       MAX9860_PGAM_SHIFT, MAX9860_PGAM_MIN, 1, pgam_tlv),
+SOC_SINGLE_TLV("Mic Preamp Volume", MAX9860_MICGAIN,
+	       MAX9860_PAM_SHIFT, MAX9860_PAM_MAX, 0, pam_tlv),
+
+SOC_ENUM("ADC Filter", avflt_enum),
+SOC_ENUM("DAC Filter", dvflt_enum),
+};
+
+static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("MICL"),
+SND_SOC_DAPM_INPUT("MICR"),
+
+SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0),
+SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0),
+
+SND_SOC_DAPM_OUTPUT("OUT"),
+
+SND_SOC_DAPM_SUPPLY("Supply", SND_SOC_NOPM, 0, 0,
+		    NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DVDD", 0, 0),
+SND_SOC_DAPM_CLOCK_SUPPLY("mclk"),
+};
+
+static const struct snd_soc_dapm_route max9860_dapm_routes[] = {
+	{ "ADCL", NULL, "MICL" },
+	{ "ADCR", NULL, "MICR" },
+	{ "AIFOUTL", NULL, "ADCL" },
+	{ "AIFOUTR", NULL, "ADCR" },
+
+	{ "DAC", NULL, "AIFINL" },
+	{ "DAC", NULL, "AIFINR" },
+	{ "OUT", NULL, "DAC" },
+
+	{ "Supply", NULL, "AVDD" },
+	{ "Supply", NULL, "DVDD" },
+	{ "Supply", NULL, "mclk" },
+
+	{ "DAC", NULL, "Supply" },
+	{ "ADCL", NULL, "Supply" },
+	{ "ADCR", NULL, "Supply" },
+};
+
+static int max9860_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 max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec);
+	u8 master;
+	u8 ifc1a = 0;
+	u8 ifc1b = 0;
+	u8 sysclk = 0;
+	unsigned long n;
+	int ret;
+
+	dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n",
+		params_rate(params),
+		params_channels(params));
+
+	if (params_channels(params) == 2)
+		ifc1b |= MAX9860_ST;
+
+	switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		master = 0;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		master = MAX9860_MASTER;
+		break;
+	default:
+		return -EINVAL;
+	}
+	ifc1a |= master;
+
+	if (master) {
+		if (params_width(params) * params_channels(params) > 48)
+			ifc1b |= MAX9860_BSEL_64X;
+		else
+			ifc1b |= MAX9860_BSEL_48X;
+	}
+
+	switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		ifc1a |= MAX9860_DDLY;
+		ifc1b |= MAX9860_ADLY;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		ifc1a |= MAX9860_WCI;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		if (params_width(params) != 16) {
+			dev_err(codec->dev,
+				"DSP_A works for 16 bits per sample only.\n");
+			return -EINVAL;
+		}
+		ifc1a |= MAX9860_DDLY | MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM;
+		ifc1b |= MAX9860_ADLY;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		if (params_width(params) != 16) {
+			dev_err(codec->dev,
+				"DSP_B works for 16 bits per sample only.\n");
+			return -EINVAL;
+		}
+		ifc1a |= MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (max9860->fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+		case SND_SOC_DAIFMT_DSP_A:
+		case SND_SOC_DAIFMT_DSP_B:
+			return -EINVAL;
+		}
+		ifc1a ^= MAX9860_WCI;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+		case SND_SOC_DAIFMT_DSP_A:
+		case SND_SOC_DAIFMT_DSP_B:
+			return -EINVAL;
+		}
+		ifc1a ^= MAX9860_WCI;
+		/* fall through */
+	case SND_SOC_DAIFMT_IB_NF:
+		ifc1a ^= MAX9860_DBCI;
+		ifc1b ^= MAX9860_ABCI;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	dev_dbg(codec->dev, "IFC1A  %02x\n", ifc1a);
+	ret = regmap_write(max9860->regmap, MAX9860_IFC1A, ifc1a);
+	if (ret) {
+		dev_err(codec->dev, "Failed to set IFC1A: %d\n", ret);
+		return ret;
+	}
+	dev_dbg(codec->dev, "IFC1B  %02x\n", ifc1b);
+	ret = regmap_write(max9860->regmap, MAX9860_IFC1B, ifc1b);
+	if (ret) {
+		dev_err(codec->dev, "Failed to set IFC1B: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * Check if Integer Clock Mode is possible, but avoid it in slave mode
+	 * since we then do not know if lrclk is derived from pclk and the
+	 * datasheet mentions that the frequencies have to match exactly in
+	 * order for this to work.
+	 */
+	if (params_rate(params) == 8000 || params_rate(params) == 16000) {
+		if (master) {
+			switch (max9860->pclk_rate) {
+			case 12000000:
+				sysclk = MAX9860_FREQ_12MHZ;
+				break;
+			case 13000000:
+				sysclk = MAX9860_FREQ_13MHZ;
+				break;
+			case 19200000:
+				sysclk = MAX9860_FREQ_19_2MHZ;
+				break;
+			default:
+				/*
+				 * Integer Clock Mode not possible. Leave
+				 * sysclk at zero and fall through to the
+				 * code below for PLL mode.
+				 */
+				break;
+			}
+
+			if (sysclk && params_rate(params) == 16000)
+				sysclk |= MAX9860_16KHZ;
+		}
+	}
+
+	/*
+	 * Largest possible n:
+	 *    65536 * 96 * 48kHz / 10MHz -> 30199
+	 * Smallest possible n:
+	 *    65536 * 96 *  8kHz / 20MHz -> 2517
+	 * Both fit nicely in the available 15 bits, no need to apply any mask.
+	 */
+	n = DIV_ROUND_CLOSEST_ULL(65536ULL * 96 * params_rate(params),
+				  max9860->pclk_rate);
+
+	if (!sysclk) {
+		/* PLL mode */
+		if (params_rate(params) > 24000)
+			sysclk |= MAX9860_16KHZ;
+
+		if (!master)
+			n |= 1; /* trigger rapid pll lock mode */
+	}
+
+	sysclk |= max9860->psclk;
+	dev_dbg(codec->dev, "SYSCLK %02x\n", sysclk);
+	ret = regmap_write(max9860->regmap,
+			   MAX9860_SYSCLK, sysclk);
+	if (ret) {
+		dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret);
+		return ret;
+	}
+	dev_dbg(codec->dev, "N %lu\n", n);
+	ret = regmap_write(max9860->regmap,
+			   MAX9860_AUDIOCLKHIGH, n >> 8);
+	if (ret) {
+		dev_err(codec->dev, "Failed to set NHI: %d\n", ret);
+		return ret;
+	}
+	ret = regmap_write(max9860->regmap,
+			   MAX9860_AUDIOCLKLOW, n & 0xff);
+	if (ret) {
+		dev_err(codec->dev, "Failed to set NLO: %d\n", ret);
+		return ret;
+	}
+
+	if (!master) {
+		dev_dbg(codec->dev, "Enable PLL\n");
+		ret = regmap_update_bits(max9860->regmap, MAX9860_AUDIOCLKHIGH,
+					 MAX9860_PLL, MAX9860_PLL);
+		if (ret) {
+			dev_err(codec->dev, "Failed to enable PLL: %d\n", ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_CBS_CFS:
+		max9860->fmt = fmt;
+		return 0;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static const struct snd_soc_dai_ops max9860_dai_ops = {
+	.hw_params = max9860_hw_params,
+	.set_fmt = max9860_set_fmt,
+};
+
+static struct snd_soc_dai_driver max9860_dai = {
+	.name = "max9860-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 8000,
+		.rate_max = 48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			   SNDRV_PCM_FMTBIT_S24_LE |
+			   SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 8000,
+		.rate_max = 48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			   SNDRV_PCM_FMTBIT_S24_LE |
+			   SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.ops = &max9860_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static int max9860_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	struct max9860_priv *max9860 = dev_get_drvdata(codec->dev);
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN,
+					 MAX9860_SHDN, MAX9860_SHDN);
+		if (ret) {
+			dev_err(codec->dev, "Failed to remove SHDN: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN,
+					 MAX9860_SHDN, 0);
+		if (ret) {
+			dev_err(codec->dev, "Failed to request SHDN: %d\n",
+				ret);
+			return ret;
+		}
+		break;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver max9860_codec_driver = {
+	.set_bias_level = max9860_set_bias_level,
+	.idle_bias_off = true,
+
+	.controls = max9860_controls,
+	.num_controls = ARRAY_SIZE(max9860_controls),
+	.dapm_widgets = max9860_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(max9860_dapm_widgets),
+	.dapm_routes = max9860_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes),
+};
+
+#ifdef CONFIG_PM
+static int max9860_suspend(struct device *dev)
+{
+	struct max9860_priv *max9860 = dev_get_drvdata(dev);
+	int ret;
+
+	ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK,
+				 MAX9860_PSCLK, MAX9860_PSCLK_OFF);
+	if (ret) {
+		dev_err(dev, "Failed to disable clock: %d\n", ret);
+		return ret;
+	}
+
+	regulator_disable(max9860->dvddio);
+
+	return 0;
+}
+
+static int max9860_resume(struct device *dev)
+{
+	struct max9860_priv *max9860 = dev_get_drvdata(dev);
+	int ret;
+
+	ret = regulator_enable(max9860->dvddio);
+	if (ret) {
+		dev_err(dev, "Failed to enable DVDDIO: %d\n", ret);
+		return ret;
+	}
+
+	regcache_cache_only(max9860->regmap, false);
+	ret = regcache_sync(max9860->regmap);
+	if (ret) {
+		dev_err(dev, "Failed to sync cache: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK,
+				 MAX9860_PSCLK, max9860->psclk);
+	if (ret) {
+		dev_err(dev, "Failed to enable clock: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+#endif
+
+const struct dev_pm_ops max9860_pm_ops = {
+	SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL)
+};
+
+static int max9860_probe(struct i2c_client *i2c,
+			 const struct i2c_device_id *id)
+{
+	struct device *dev = &i2c->dev;
+	struct max9860_priv *max9860;
+	int ret;
+	struct clk *mclk;
+	unsigned long mclk_rate;
+	int i;
+	int intr;
+
+	max9860 = devm_kzalloc(dev, sizeof(struct max9860_priv), GFP_KERNEL);
+	if (!max9860)
+		return -ENOMEM;
+
+	max9860->dvddio = devm_regulator_get(dev, "DVDDIO");
+	if (IS_ERR(max9860->dvddio)) {
+		ret = PTR_ERR(max9860->dvddio);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get DVDDIO supply: %d\n", ret);
+		return ret;
+	}
+
+	max9860->dvddio_nb.notifier_call = max9860_dvddio_event;
+
+	ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb);
+	if (ret)
+		dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret);
+
+	ret = regulator_enable(max9860->dvddio);
+	if (ret != 0) {
+		dev_err(dev, "Failed to enable DVDDIO: %d\n", ret);
+		return ret;
+	}
+
+	max9860->regmap = devm_regmap_init_i2c(i2c, &max9860_regmap);
+	if (IS_ERR(max9860->regmap)) {
+		ret = PTR_ERR(max9860->regmap);
+		goto err_regulator;
+	}
+
+	dev_set_drvdata(dev, max9860);
+
+	/*
+	 * mclk has to be in the 10MHz to 60MHz range.
+	 * psclk is used to scale mclk into pclk so that
+	 * pclk is in the 10MHz to 20MHz range.
+	 */
+	mclk = clk_get(dev, "mclk");
+
+	if (IS_ERR(mclk)) {
+		ret = PTR_ERR(mclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "Failed to get MCLK: %d\n", ret);
+		goto err_regulator;
+	}
+
+	mclk_rate = clk_get_rate(mclk);
+	clk_put(mclk);
+
+	if (mclk_rate > 60000000 || mclk_rate < 10000000) {
+		dev_err(dev, "Bad mclk %luHz (needs 10MHz - 60MHz)\n",
+			mclk_rate);
+		ret = -EINVAL;
+		goto err_regulator;
+	}
+	if (mclk_rate >= 40000000)
+		max9860->psclk = 3;
+	else if (mclk_rate >= 20000000)
+		max9860->psclk = 2;
+	else
+		max9860->psclk = 1;
+	max9860->pclk_rate = mclk_rate >> (max9860->psclk - 1);
+	max9860->psclk <<= MAX9860_PSCLK_SHIFT;
+	dev_dbg(dev, "mclk %lu pclk %lu\n", mclk_rate, max9860->pclk_rate);
+
+	regcache_cache_bypass(max9860->regmap, true);
+	for (i = 0; i < max9860_regmap.num_reg_defaults; ++i) {
+		ret = regmap_write(max9860->regmap,
+				   max9860_regmap.reg_defaults[i].reg,
+				   max9860_regmap.reg_defaults[i].def);
+		if (ret) {
+			dev_err(dev, "Failed to initialize register %u: %d\n",
+				max9860_regmap.reg_defaults[i].reg, ret);
+			goto err_regulator;
+		}
+	}
+	regcache_cache_bypass(max9860->regmap, false);
+
+	ret = regmap_read(max9860->regmap, MAX9860_INTRSTATUS, &intr);
+	if (ret) {
+		dev_err(dev, "Failed to clear INTRSTATUS: %d\n", ret);
+		goto err_regulator;
+	}
+
+	pm_runtime_set_active(dev);
+	pm_runtime_enable(dev);
+	pm_runtime_idle(dev);
+
+	ret = snd_soc_register_codec(dev, &max9860_codec_driver,
+				     &max9860_dai, 1);
+	if (ret) {
+		dev_err(dev, "Failed to register CODEC: %d\n", ret);
+		goto err_pm;
+	}
+
+	return 0;
+
+err_pm:
+	pm_runtime_disable(dev);
+err_regulator:
+	regulator_disable(max9860->dvddio);
+	return ret;
+}
+
+static int max9860_remove(struct i2c_client *i2c)
+{
+	struct device *dev = &i2c->dev;
+	struct max9860_priv *max9860 = dev_get_drvdata(dev);
+
+	snd_soc_unregister_codec(dev);
+	pm_runtime_disable(dev);
+	regulator_disable(max9860->dvddio);
+	return 0;
+}
+
+static const struct i2c_device_id max9860_i2c_id[] = {
+	{ "max9860", },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, max9860_i2c_id);
+
+static const struct of_device_id max9860_of_match[] = {
+	{ .compatible = "maxim,max9860", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, max9860_of_match);
+
+static struct i2c_driver max9860_i2c_driver = {
+	.probe	        = max9860_probe,
+	.remove         = max9860_remove,
+	.id_table       = max9860_i2c_id,
+	.driver         = {
+		.name           = "max9860",
+		.of_match_table = max9860_of_match,
+		.pm             = &max9860_pm_ops,
+	},
+};
+
+module_i2c_driver(max9860_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC MAX9860 Mono Audio Voice Codec driver");
+MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/max9860.h b/sound/soc/codecs/max9860.h
new file mode 100644
index 000000000000..22041bd67a7d
--- /dev/null
+++ b/sound/soc/codecs/max9860.h
@@ -0,0 +1,162 @@
+/*
+ * Driver for the MAX9860 Mono Audio Voice Codec
+ *
+ * Author: Peter Rosin <peda@axentia.s>
+ *         Copyright 2016 Axentia Technologies
+ *
+ * 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.
+ */
+
+#ifndef _SND_SOC_MAX9860
+#define _SND_SOC_MAX9860
+
+#define MAX9860_INTRSTATUS   0x00
+#define MAX9860_MICREADBACK  0x01
+#define MAX9860_INTEN        0x02
+#define MAX9860_SYSCLK       0x03
+#define MAX9860_AUDIOCLKHIGH 0x04
+#define MAX9860_AUDIOCLKLOW  0x05
+#define MAX9860_IFC1A        0x06
+#define MAX9860_IFC1B        0x07
+#define MAX9860_VOICEFLTR    0x08
+#define MAX9860_DACATTN      0x09
+#define MAX9860_ADCLEVEL     0x0a
+#define MAX9860_DACGAIN      0x0b
+#define MAX9860_MICGAIN      0x0c
+#define MAX9860_RESERVED     0x0d
+#define MAX9860_MICADC       0x0e
+#define MAX9860_NOISEGATE    0x0f
+#define MAX9860_PWRMAN       0x10
+#define MAX9860_REVISION     0xff
+
+#define MAX9860_MAX_REGISTER 0xff
+
+/* INTRSTATUS */
+#define MAX9860_CLD          0x80
+#define MAX9860_SLD          0x40
+#define MAX9860_ULK          0x20
+
+/* MICREADBACK */
+#define MAX9860_NG           0xe0
+#define MAX9860_AGC          0x1f
+
+/* INTEN */
+#define MAX9860_ICLD         0x80
+#define MAX9860_ISLD         0x40
+#define MAX9860_IULK         0x20
+
+/* SYSCLK */
+#define MAX9860_PSCLK        0x30
+#define MAX9860_PSCLK_OFF    0x00
+#define MAX9860_PSCLK_SHIFT  4
+#define MAX9860_FREQ         0x06
+#define MAX9860_FREQ_NORMAL  0x00
+#define MAX9860_FREQ_12MHZ   0x02
+#define MAX9860_FREQ_13MHZ   0x04
+#define MAX9860_FREQ_19_2MHZ 0x06
+#define MAX9860_16KHZ        0x01
+
+/* AUDIOCLKHIGH */
+#define MAX9860_PLL          0x80
+#define MAX9860_NHI          0x7f
+
+/* AUDIOCLKLOW */
+#define MAX9860_NLO          0xff
+
+/* IFC1A */
+#define MAX9860_MASTER       0x80
+#define MAX9860_WCI          0x40
+#define MAX9860_DBCI         0x20
+#define MAX9860_DDLY         0x10
+#define MAX9860_HIZ          0x08
+#define MAX9860_TDM          0x04
+
+/* IFC1B */
+#define MAX9860_ABCI         0x20
+#define MAX9860_ADLY         0x10
+#define MAX9860_ST           0x08
+#define MAX9860_BSEL         0x07
+#define MAX9860_BSEL_OFF     0x00
+#define MAX9860_BSEL_64X     0x01
+#define MAX9860_BSEL_48X     0x02
+#define MAX9860_BSEL_PCLK_2  0x04
+#define MAX9860_BSEL_PCLK_4  0x05
+#define MAX9860_BSEL_PCLK_8  0x06
+#define MAX9860_BSEL_PCLK_16 0x07
+
+/* VOICEFLTR */
+#define MAX9860_AVFLT        0xf0
+#define MAX9860_AVFLT_SHIFT  4
+#define MAX9860_AVFLT_COUNT  6
+#define MAX9860_DVFLT        0x0f
+#define MAX9860_DVFLT_SHIFT  0
+#define MAX9860_DVFLT_COUNT  6
+
+/* DACATTN */
+#define MAX9860_DVA          0xfe
+#define MAX9860_DVA_SHIFT    1
+#define MAX9860_DVA_MUTE     0x5e
+
+/* ADCLEVEL */
+#define MAX9860_ADCRL        0xf0
+#define MAX9860_ADCRL_SHIFT  4
+#define MAX9860_ADCLL        0x0f
+#define MAX9860_ADCLL_SHIFT  0
+#define MAX9860_ADCxL_MIN    15
+
+/* DACGAIN */
+#define MAX9860_DVG          0x60
+#define MAX9860_DVG_SHIFT    5
+#define MAX9860_DVG_MAX      3
+#define MAX9860_DVST         0x1f
+#define MAX9860_DVST_SHIFT   0
+#define MAX9860_DVST_MIN     31
+
+/* MICGAIN */
+#define MAX9860_PAM          0x60
+#define MAX9860_PAM_SHIFT    5
+#define MAX9860_PAM_MAX      3
+#define MAX9860_PGAM         0x1f
+#define MAX9860_PGAM_SHIFT   0
+#define MAX9860_PGAM_MIN     20
+
+/* MICADC */
+#define MAX9860_AGCSRC       0x80
+#define MAX9860_AGCSRC_SHIFT 7
+#define MAX9860_AGCSRC_COUNT 2
+#define MAX9860_AGCRLS       0x70
+#define MAX9860_AGCRLS_SHIFT 4
+#define MAX9860_AGCRLS_COUNT 8
+#define MAX9860_AGCATK       0x0c
+#define MAX9860_AGCATK_SHIFT 2
+#define MAX9860_AGCATK_COUNT 4
+#define MAX9860_AGCHLD       0x03
+#define MAX9860_AGCHLD_OFF   0x00
+#define MAX9860_AGCHLD_SHIFT 0
+#define MAX9860_AGCHLD_COUNT 4
+
+/* NOISEGATE */
+#define MAX9860_ANTH         0xf0
+#define MAX9860_ANTH_SHIFT   4
+#define MAX9860_ANTH_MAX     15
+#define MAX9860_AGCTH        0x0f
+#define MAX9860_AGCTH_SHIFT  0
+#define MAX9860_AGCTH_MIN    15
+
+/* PWRMAN */
+#define MAX9860_SHDN         0x80
+#define MAX9860_DACEN        0x08
+#define MAX9860_DACEN_SHIFT  3
+#define MAX9860_ADCLEN       0x02
+#define MAX9860_ADCLEN_SHIFT 1
+#define MAX9860_ADCREN       0x01
+#define MAX9860_ADCREN_SHIFT 0
+
+#endif /* _SND_SOC_MAX9860 */
-- 
2.1.4

[PATCH v4 3/4] ASoC: dapm: support mixer controls with mute at non-zero value

[Peter Rosin]

The max9860 codec has a mixer control field that has its mute/disable at
the wrong end of the scale. I.e. you turn the volume up and up, and then
as the final step the volume is off. This does not sit well with DAPM,
which assumes the mute/off is at the minimum value.

Add support for such backwards controls with code that searches TLV ranges
for the mute value and use that as trigger for DAPM off.

Signed-off-by: Peter Rosin <peda@axentia.se>
---
 sound/soc/soc-dapm.c | 38 +++++++++++++++++++++++++++++++++++---
 1 file changed, 35 insertions(+), 3 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c4464858bf01..8a1131781339 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -42,6 +42,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/initval.h>
+#include <sound/tlv.h>
 
 #include <trace/events/asoc.h>
 
@@ -722,16 +723,46 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
 	return -ENODEV;
 }
 
+static int dapm_find_tlv_mute(const unsigned int *tlv)
+{
+	int cnt;
+	const unsigned int *range;
+
+	if (!tlv || tlv[0] != SNDRV_CTL_TLVT_DB_RANGE)
+		return 0;
+
+	cnt = tlv[1] / sizeof(unsigned int);
+
+	/*
+	 * Each group of six values should be
+	 * { start end type len min step/mute }
+	 */
+	for (range = &tlv[2]; cnt >= 6; cnt -= 6, range += 6) {
+		if (range[2] != SNDRV_CTL_TLVT_DB_SCALE)
+			return 0; /* wrong type, terminate */
+		if (range[3] != 2 * sizeof(unsigned int))
+			return 0; /* wrong len, terminate */
+		if (!(range[5] & TLV_DB_SCALE_MUTE))
+			continue; /* no mute in this range */
+		return range[0]; /* start of this range is the mute value */
+	}
+
+	return 0;
+}
+
 /* set up initial codec paths */
 static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i)
 {
+	const struct snd_kcontrol_new *kcontrol_new
+		= &p->sink->kcontrol_news[i];
 	struct soc_mixer_control *mc = (struct soc_mixer_control *)
-		p->sink->kcontrol_news[i].private_value;
+		kcontrol_new->private_value;
 	unsigned int reg = mc->reg;
 	unsigned int shift = mc->shift;
 	unsigned int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
+	int mute_value = dapm_find_tlv_mute(kcontrol_new->tlv.p);
 	unsigned int val;
 
 	if (reg != SND_SOC_NOPM) {
@@ -739,7 +770,7 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i)
 		val = (val >> shift) & mask;
 		if (invert)
 			val = max - val;
-		p->connect = !!val;
+		p->connect = val != mute_value;
 	} else {
 		p->connect = 0;
 	}
@@ -3045,6 +3076,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
+	int mute_value = dapm_find_tlv_mute(kcontrol->tlv.p);
 	unsigned int val;
 	int connect, change, reg_change = 0;
 	struct snd_soc_dapm_update update;
@@ -3056,7 +3088,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 			 kcontrol->id.name);
 
 	val = (ucontrol->value.integer.value[0] & mask);
-	connect = !!val;
+	connect = val != mute_value;
 
 	if (invert)
 		val = max - val;
-- 
2.1.4

[PATCH v4 4/4] ASoC: MAX9860: add sidetone mixer control

[Peter Rosin]

Signed-off-by: Peter Rosin <peda@axentia.se>
---
 sound/soc/codecs/max9860.c | 22 +++++++++++++++++-----
 1 file changed, 17 insertions(+), 5 deletions(-)

diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c
index 2b0dd6a18dad..8d257cc20ef7 100644
--- a/sound/soc/codecs/max9860.c
+++ b/sound/soc/codecs/max9860.c
@@ -3,9 +3,6 @@
  *
  * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf
  *
- * The driver does not support sidetone since the DVST register field is
- * backwards with the mute near the maximum level instead of the minimum.
- *
  * Author: Peter Rosin <peda@axentia.s>
  *         Copyright 2016 Axentia Technologies
  *
@@ -135,6 +132,10 @@ const struct regmap_config max9860_regmap = {
 static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1);
 static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0);
 static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0);
+/* The dvst field has its mute in the wrong end. Sigh. */
+static const DECLARE_TLV_DB_RANGE(dvst_tlv,
+	0, MAX9860_DVST_MIN - 1,            TLV_DB_SCALE_ITEM(-6000, 200, 0),
+	MAX9860_DVST_MIN, MAX9860_DVST_MIN, TLV_DB_SCALE_ITEM(0, 0, 1));
 static const DECLARE_TLV_DB_RANGE(pam_tlv,
 	0, MAX9860_PAM_MAX - 1,             TLV_DB_SCALE_ITEM(-2000, 2000, 1),
 	MAX9860_PAM_MAX, MAX9860_PAM_MAX,   TLV_DB_SCALE_ITEM(3000, 0, 0));
@@ -214,6 +215,11 @@ SOC_ENUM("ADC Filter", avflt_enum),
 SOC_ENUM("DAC Filter", dvflt_enum),
 };
 
+static const struct snd_kcontrol_new max9860_mixer_controls[] = {
+SOC_DAPM_SINGLE_TLV("Sidetone Volume", MAX9860_DACGAIN,
+		    MAX9860_DVST_SHIFT, MAX9860_DVST_MIN, 1, dvst_tlv),
+};
+
 static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("MICL"),
 SND_SOC_DAPM_INPUT("MICR"),
@@ -224,6 +230,10 @@ SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0),
 SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0),
 
+SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 0,
+		   max9860_mixer_controls,
+		   ARRAY_SIZE(max9860_mixer_controls)),
+
 SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0),
 
@@ -244,8 +254,10 @@ static const struct snd_soc_dapm_route max9860_dapm_routes[] = {
 	{ "AIFOUTL", NULL, "ADCL" },
 	{ "AIFOUTR", NULL, "ADCR" },
 
-	{ "DAC", NULL, "AIFINL" },
-	{ "DAC", NULL, "AIFINR" },
+	{ "Mixer", NULL, "AIFINL" },
+	{ "Mixer", NULL, "AIFINR" },
+	{ "Mixer", "Sidetone Volume", "ADCL" },
+	{ "DAC", NULL, "Mixer" },
 	{ "OUT", NULL, "DAC" },
 
 	{ "Supply", NULL, "AVDD" },
-- 
2.1.4

Re: [PATCH v4 1/4] dt-bindings: sound: add bindings for the max9860 codec

[Mark Brown]

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On Sat, May 14, 2016 at 11:09:38PM +0200, Peter Rosin wrote:
> This adds the device tree binding documentation for the Maxim Integrated
> MAX9860 mono audio voice codec.

Please submit patches using subject lines reflecting the style for the
subsystem.  This makes it easier for people to identify relevant
patches.

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Re: [PATCH v4 1/4] dt-bindings: sound: add bindings for the max9860 codec

[Peter Rosin]

On 2016-05-27 22:24, Mark Brown wrote:
> On Sat, May 14, 2016 at 11:09:38PM +0200, Peter Rosin wrote:
>> This adds the device tree binding documentation for the Maxim Integrated
>> MAX9860 mono audio voice codec.
> 
> Please submit patches using subject lines reflecting the style for the
> subsystem.  This makes it easier for people to identify relevant
> patches.

Ok, but to me it was obviously not obvious if I submitted this patch to
ASoC or if I submitted it to devicetree.

Should I resend over this detail or will you fix it up?

Cheers,
Peter

Re: [PATCH v4 1/4] dt-bindings: sound: add bindings for the max9860 codec

[Mark Brown]

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On Fri, May 27, 2016 at 10:40:07PM +0200, Peter Rosin wrote:

> Ok, but to me it was obviously not obvious if I submitted this patch to
> ASoC or if I submitted it to devicetree.

> Should I resend over this detail or will you fix it up?

It's OK, I've applied it already.

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Re: [PATCH v4 1/4] dt-bindings: sound: add bindings for the max9860 codec

[Peter Rosin]

On 2016-05-27 22:45, Mark Brown wrote:
> On Fri, May 27, 2016 at 10:40:07PM +0200, Peter Rosin wrote:
> 
>> Ok, but to me it was obviously not obvious if I submitted this patch to
>> ASoC or if I submitted it to devicetree.
> 
>> Should I resend over this detail or will you fix it up?
> 
> It's OK, I've applied it already.

Great, thanks!

*me peeks at the .../sound.git repo*

You have named the topic branch topic/max8960, which should be
topic/max9860. I guess wm89.. is hard-wired into your fingers :-)

(In case it matters)

Cheers,
Peter

Re: [PATCH v4 3/4] ASoC: dapm: support mixer controls with mute at non-zero value

[Peter Rosin]

Hi!

Sorry to send a ping like this, with the patch still in patchwork
and all. But it's been a month since 1/4 and 2/4 were committed
and I expected at least some comment on the approach for patches
3 and 4...

Cheers,
Peter

On 2016-05-14 23:09, Peter Rosin wrote:
> The max9860 codec has a mixer control field that has its mute/disable at
> the wrong end of the scale. I.e. you turn the volume up and up, and then
> as the final step the volume is off. This does not sit well with DAPM,
> which assumes the mute/off is at the minimum value.
> 
> Add support for such backwards controls with code that searches TLV ranges
> for the mute value and use that as trigger for DAPM off.
> 
> Signed-off-by: Peter Rosin <peda@axentia.se>
> ---
>  sound/soc/soc-dapm.c | 38 +++++++++++++++++++++++++++++++++++---
>  1 file changed, 35 insertions(+), 3 deletions(-)
> 
> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
> index c4464858bf01..8a1131781339 100644
> --- a/sound/soc/soc-dapm.c
> +++ b/sound/soc/soc-dapm.c
> @@ -42,6 +42,7 @@
>  #include <sound/pcm_params.h>
>  #include <sound/soc.h>
>  #include <sound/initval.h>
> +#include <sound/tlv.h>
>  
>  #include <trace/events/asoc.h>
>  
> @@ -722,16 +723,46 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
>  	return -ENODEV;
>  }
>  
> +static int dapm_find_tlv_mute(const unsigned int *tlv)
> +{
> +	int cnt;
> +	const unsigned int *range;
> +
> +	if (!tlv || tlv[0] != SNDRV_CTL_TLVT_DB_RANGE)
> +		return 0;
> +
> +	cnt = tlv[1] / sizeof(unsigned int);
> +
> +	/*
> +	 * Each group of six values should be
> +	 * { start end type len min step/mute }
> +	 */
> +	for (range = &tlv[2]; cnt >= 6; cnt -= 6, range += 6) {
> +		if (range[2] != SNDRV_CTL_TLVT_DB_SCALE)
> +			return 0; /* wrong type, terminate */
> +		if (range[3] != 2 * sizeof(unsigned int))
> +			return 0; /* wrong len, terminate */
> +		if (!(range[5] & TLV_DB_SCALE_MUTE))
> +			continue; /* no mute in this range */
> +		return range[0]; /* start of this range is the mute value */
> +	}
> +
> +	return 0;
> +}
> +
>  /* set up initial codec paths */
>  static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i)
>  {
> +	const struct snd_kcontrol_new *kcontrol_new
> +		= &p->sink->kcontrol_news[i];
>  	struct soc_mixer_control *mc = (struct soc_mixer_control *)
> -		p->sink->kcontrol_news[i].private_value;
> +		kcontrol_new->private_value;
>  	unsigned int reg = mc->reg;
>  	unsigned int shift = mc->shift;
>  	unsigned int max = mc->max;
>  	unsigned int mask = (1 << fls(max)) - 1;
>  	unsigned int invert = mc->invert;
> +	int mute_value = dapm_find_tlv_mute(kcontrol_new->tlv.p);
>  	unsigned int val;
>  
>  	if (reg != SND_SOC_NOPM) {
> @@ -739,7 +770,7 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i)
>  		val = (val >> shift) & mask;
>  		if (invert)
>  			val = max - val;
> -		p->connect = !!val;
> +		p->connect = val != mute_value;
>  	} else {
>  		p->connect = 0;
>  	}
> @@ -3045,6 +3076,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
>  	int max = mc->max;
>  	unsigned int mask = (1 << fls(max)) - 1;
>  	unsigned int invert = mc->invert;
> +	int mute_value = dapm_find_tlv_mute(kcontrol->tlv.p);
>  	unsigned int val;
>  	int connect, change, reg_change = 0;
>  	struct snd_soc_dapm_update update;
> @@ -3056,7 +3088,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
>  			 kcontrol->id.name);
>  
>  	val = (ucontrol->value.integer.value[0] & mask);
> -	connect = !!val;
> +	connect = val != mute_value;
>  
>  	if (invert)
>  		val = max - val;
> 

Re: [PATCH v4 3/4] ASoC: dapm: support mixer controls with mute at non-zero value

[Mark Brown]

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On Tue, Jun 28, 2016 at 10:12:37AM +0200, Peter Rosin wrote:
> Hi!
> 
> Sorry to send a ping like this, with the patch still in patchwork
> and all. But it's been a month since 1/4 and 2/4 were committed
> and I expected at least some comment on the approach for patches
> 3 and 4...

Please don't send content free pings and please allow a reasonable time
for review.  People get busy, go on holiday, attend conferences and so 
on so unless there is some reason for urgency (like critical bug fixes)
please allow at least a couple of weeks for review.  Sending content
free pings just adds to the mail volume (if they are seen at all) and if 
something has gone wrong you'll have to resend the patches anyway.

Please don't top post, reply in line with needed context.  This allows
readers to readily follow the flow of conversation and understand what
you are talking about and also helps ensure that everything in the
discussion is being addressed.

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