Message-Id: <20230614074904.29085-13-herve.codina@bootlin.com>
Date:   Wed, 14 Jun 2023 09:49:03 +0200
From:   Herve Codina <herve.codina@...tlin.com>
To:     Herve Codina <herve.codina@...tlin.com>,
        Liam Girdwood <lgirdwood@...il.com>,
        Mark Brown <broonie@...nel.org>,
        Rob Herring <robh+dt@...nel.org>,
        Krzysztof Kozlowski <krzysztof.kozlowski+dt@...aro.org>,
        Conor Dooley <conor+dt@...nel.org>,
        Jonathan Cameron <jic23@...nel.org>,
        Lars-Peter Clausen <lars@...afoo.de>,
        Jaroslav Kysela <perex@...ex.cz>,
        Takashi Iwai <tiwai@...e.com>,
        Kuninori Morimoto <kuninori.morimoto.gx@...esas.com>,
        Andy Shevchenko <andy.shevchenko@...il.com>
Cc:     alsa-devel@...a-project.org, devicetree@...r.kernel.org,
        linux-kernel@...r.kernel.org, linux-iio@...r.kernel.org,
        Christophe Leroy <christophe.leroy@...roup.eu>,
        Thomas Petazzoni <thomas.petazzoni@...tlin.com>
Subject: [PATCH v4 12/13] ASoC: codecs: Add support for the generic IIO auxiliary devices

Industrial I/O devices can be present in the audio path.
These devices needs to be used as audio components in order to be
fully integrated in the audio path.

This support allows to consider these Industrial I/O devices as
auxiliary audio devices and allows one to control them using mixer
controls.

Signed-off-by: Herve Codina <herve.codina@...tlin.com>
---
 sound/soc/codecs/Kconfig         |  12 ++
 sound/soc/codecs/Makefile        |   2 +
 sound/soc/codecs/audio-iio-aux.c | 338 +++++++++++++++++++++++++++++++
 3 files changed, 352 insertions(+)
 create mode 100644 sound/soc/codecs/audio-iio-aux.c

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 44806bfe8ee5..92b7c417f1b2 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_AK5558
 	imply SND_SOC_ALC5623
 	imply SND_SOC_ALC5632
+	imply SND_SOC_AUDIO_IIO_AUX
 	imply SND_SOC_AW8738
 	imply SND_SOC_AW88395
 	imply SND_SOC_BT_SCO
@@ -608,6 +609,17 @@ config SND_SOC_ALC5632
 	tristate
 	depends on I2C
 
+config SND_SOC_AUDIO_IIO_AUX
+	tristate "Audio IIO Auxiliary device"
+	depends on IIO
+	help
+	  Enable support for Industrial I/O devices as audio auxiliary devices.
+	  This allows to have an IIO device present in the audio path and
+	  controlled using mixer controls.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-soc-audio-iio-aux.
+
 config SND_SOC_AW8738
 	tristate "Awinic AW8738 Audio Amplifier"
 	select GPIOLIB
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 2c45c2f97e4e..f2828d3616c5 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -45,6 +45,7 @@ snd-soc-ak4671-objs := ak4671.o
 snd-soc-ak5386-objs := ak5386.o
 snd-soc-ak5558-objs := ak5558.o
 snd-soc-arizona-objs := arizona.o arizona-jack.o
+snd-soc-audio-iio-aux-objs := audio-iio-aux.o
 snd-soc-aw8738-objs := aw8738.o
 snd-soc-aw88395-lib-objs := aw88395/aw88395_lib.o
 snd-soc-aw88395-objs := aw88395/aw88395.o \
@@ -421,6 +422,7 @@ obj-$(CONFIG_SND_SOC_AK5558)	+= snd-soc-ak5558.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
 obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
+obj-$(CONFIG_SND_SOC_AUDIO_IIO_AUX)	+= snd-soc-audio-iio-aux.o
 obj-$(CONFIG_SND_SOC_AW8738)	+= snd-soc-aw8738.o
 obj-$(CONFIG_SND_SOC_AW88395_LIB) += snd-soc-aw88395-lib.o
 obj-$(CONFIG_SND_SOC_AW88395)	+=snd-soc-aw88395.o
diff --git a/sound/soc/codecs/audio-iio-aux.c b/sound/soc/codecs/audio-iio-aux.c
new file mode 100644
index 000000000000..b9d72cbb85f2
--- /dev/null
+++ b/sound/soc/codecs/audio-iio-aux.c
@@ -0,0 +1,338 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// ALSA SoC glue to use IIO devices as audio components
+//
+// Copyright 2023 CS GROUP France
+//
+// Author: Herve Codina <herve.codina@...tlin.com>
+
+#include <linux/iio/consumer.h>
+#include <linux/minmax.h>
+#include <linux/mod_devicetable.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/string_helpers.h>
+
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+struct audio_iio_aux_chan {
+	struct iio_channel *iio_chan;
+	const char *name;
+	int max;
+	int min;
+	bool is_invert_range;
+};
+
+struct audio_iio_aux {
+	struct device *dev;
+	struct audio_iio_aux_chan *chans;
+	unsigned int num_chans;
+};
+
+static int audio_iio_aux_info_volsw(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_info *uinfo)
+{
+	struct audio_iio_aux_chan *chan = (struct audio_iio_aux_chan *)kcontrol->private_value;
+
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = chan->max - chan->min;
+	uinfo->type = (uinfo->value.integer.max == 1) ?
+			SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+	return 0;
+}
+
+static int audio_iio_aux_get_volsw(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct audio_iio_aux_chan *chan = (struct audio_iio_aux_chan *)kcontrol->private_value;
+	int max = chan->max;
+	int min = chan->min;
+	bool invert_range = chan->is_invert_range;
+	int ret;
+	int val;
+
+	ret = iio_read_channel_raw(chan->iio_chan, &val);
+	if (ret < 0)
+		return ret;
+
+	ucontrol->value.integer.value[0] = val - min;
+	if (invert_range)
+		ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0];
+
+	return 0;
+}
+
+static int audio_iio_aux_put_volsw(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct audio_iio_aux_chan *chan = (struct audio_iio_aux_chan *)kcontrol->private_value;
+	int max = chan->max;
+	int min = chan->min;
+	bool invert_range = chan->is_invert_range;
+	int val;
+	int ret;
+	int tmp;
+
+	val = ucontrol->value.integer.value[0];
+	if (val < 0)
+		return -EINVAL;
+	if (val > max - min)
+		return -EINVAL;
+
+	val = val + min;
+	if (invert_range)
+		val = max - val;
+
+	ret = iio_read_channel_raw(chan->iio_chan, &tmp);
+	if (ret < 0)
+		return ret;
+
+	if (tmp == val)
+		return 0;
+
+	ret = iio_write_channel_raw(chan->iio_chan, val);
+	if (ret)
+		return ret;
+
+	return 1; /* The value changed */
+}
+
+static int audio_iio_aux_add_controls(struct snd_soc_component *component,
+				      struct audio_iio_aux_chan *chan)
+{
+	struct snd_kcontrol_new control = {};
+
+	control.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	control.name = chan->name;
+	control.info = audio_iio_aux_info_volsw;
+	control.get = audio_iio_aux_get_volsw;
+	control.put = audio_iio_aux_put_volsw;
+	control.private_value = (unsigned long)chan;
+
+	return snd_soc_add_component_controls(component, &control, 1);
+}
+
+/*
+ * These data could be on stack but they are pretty big.
+ * As ASoC internally copy them and protect them against concurrent accesses
+ * (snd_soc_bind_card() protects using client_mutex), keep them in the global
+ * data area.
+ */
+static struct snd_soc_dapm_widget widgets[3];
+static struct snd_soc_dapm_route routes[2];
+
+/* Be sure sizes are correct (need 3 widgets and 2 routes) */
+static_assert(ARRAY_SIZE(widgets) >= 3, "3 widgets are needed");
+static_assert(ARRAY_SIZE(routes) >= 2, "2 routes are needed");
+
+static int audio_iio_aux_add_dapms(struct snd_soc_component *component,
+				   struct audio_iio_aux_chan *chan)
+{
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	char *output_name;
+	char *input_name;
+	char *pga_name;
+	int ret;
+
+	input_name = kasprintf(GFP_KERNEL, "%s IN", chan->name);
+	if (!input_name)
+		return -ENOMEM;
+
+	output_name = kasprintf(GFP_KERNEL, "%s OUT", chan->name);
+	if (!output_name) {
+		ret = -ENOMEM;
+		goto out_free_input_name;
+	}
+	pga_name = kasprintf(GFP_KERNEL, "%s PGA", chan->name);
+	if (!pga_name) {
+		ret = -ENOMEM;
+		goto out_free_output_name;
+	}
+
+	widgets[0] = SND_SOC_DAPM_INPUT(input_name);
+	widgets[1] = SND_SOC_DAPM_OUTPUT(output_name);
+	widgets[2] = SND_SOC_DAPM_PGA(pga_name, SND_SOC_NOPM, 0, 0, NULL, 0);
+	ret = snd_soc_dapm_new_controls(dapm, widgets, 3);
+	if (ret)
+		goto out_free_pga_name;
+
+	routes[0].sink = pga_name;
+	routes[0].control = NULL;
+	routes[0].source = input_name;
+	routes[1].sink = output_name;
+	routes[1].control = NULL;
+	routes[1].source = pga_name;
+	ret = snd_soc_dapm_add_routes(dapm, routes, 2);
+
+	/* Allocated names are no more needed (duplicated in ASoC internals) */
+
+out_free_pga_name:
+	kfree(pga_name);
+out_free_output_name:
+	kfree(output_name);
+out_free_input_name:
+	kfree(input_name);
+	return ret;
+}
+
+static int audio_iio_aux_component_probe(struct snd_soc_component *component)
+{
+	struct audio_iio_aux *iio_aux = snd_soc_component_get_drvdata(component);
+	struct audio_iio_aux_chan *chan;
+	int ret;
+	int i;
+
+	for (i = 0; i < iio_aux->num_chans; i++) {
+		chan = iio_aux->chans + i;
+
+		ret = iio_read_max_channel_raw(chan->iio_chan, &chan->max);
+		if (ret)
+			return dev_err_probe(component->dev, ret,
+					     "chan[%d] %s: Cannot get max raw value\n",
+					     i, chan->name);
+
+		ret = iio_read_min_channel_raw(chan->iio_chan, &chan->min);
+		if (ret)
+			return dev_err_probe(component->dev, ret,
+					     "chan[%d] %s: Cannot get min raw value\n",
+					     i, chan->name);
+
+		if (chan->min > chan->max) {
+			dev_dbg(component->dev, "chan[%d] %s: Swap min and max\n",
+				i, chan->name);
+			swap(chan->min, chan->max);
+		}
+
+		/* Set initial value */
+		ret = iio_write_channel_raw(chan->iio_chan,
+					    chan->is_invert_range ? chan->max : chan->min);
+		if (ret)
+			return dev_err_probe(component->dev, ret,
+					     "chan[%d] %s: Cannot set initial value\n",
+					     i, chan->name);
+
+		ret = audio_iio_aux_add_controls(component, chan);
+		if (ret)
+			return ret;
+
+		ret = audio_iio_aux_add_dapms(component, chan);
+		if (ret)
+			return ret;
+
+		dev_dbg(component->dev, "chan[%d]: Added %s (min=%d, max=%d, invert=%s)\n",
+			i, chan->name, chan->min, chan->max,
+			str_on_off(chan->is_invert_range));
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver audio_iio_aux_component_driver = {
+	.probe = audio_iio_aux_component_probe,
+};
+
+static int audio_iio_aux_probe(struct platform_device *pdev)
+{
+	struct audio_iio_aux_chan *iio_aux_chan;
+	struct device *dev = &pdev->dev;
+	struct audio_iio_aux *iio_aux;
+	const char **names;
+	u32 *invert_ranges;
+	int count;
+	int ret;
+	int i;
+
+	iio_aux = devm_kzalloc(dev, sizeof(*iio_aux), GFP_KERNEL);
+	if (!iio_aux)
+		return -ENOMEM;
+
+	iio_aux->dev = dev;
+
+	count = device_property_string_array_count(dev, "io-channel-names");
+	if (count < 0)
+		return dev_err_probe(dev, count, "failed to count io-channel-names\n");
+
+	iio_aux->num_chans = count;
+
+	iio_aux->chans = devm_kmalloc_array(dev, iio_aux->num_chans,
+					    sizeof(*iio_aux->chans), GFP_KERNEL);
+	if (!iio_aux->chans)
+		return -ENOMEM;
+
+	names = kcalloc(iio_aux->num_chans, sizeof(*names), GFP_KERNEL);
+	if (!names)
+		return -ENOMEM;
+
+	invert_ranges = kcalloc(iio_aux->num_chans, sizeof(*invert_ranges), GFP_KERNEL);
+	if (!invert_ranges) {
+		ret = -ENOMEM;
+		goto out_free_names;
+	}
+
+	ret = device_property_read_string_array(dev, "io-channel-names",
+						names, iio_aux->num_chans);
+	if (ret < 0) {
+		dev_err_probe(dev, ret, "failed to read io-channel-names\n");
+		goto out_free_invert_ranges;
+	}
+
+	/*
+	 * snd-control-invert-range is optional and can contain fewer items
+	 * than the number of channels. Unset values default to 0.
+	 */
+	count = device_property_count_u32(dev, "snd-control-invert-range");
+	if (count > 0) {
+		count = min_t(unsigned int, count, iio_aux->num_chans);
+		ret = device_property_read_u32_array(dev, "snd-control-invert-range",
+						     invert_ranges, count);
+		if (ret < 0) {
+			dev_err_probe(dev, ret, "failed to read snd-control-invert-range\n");
+			goto out_free_invert_ranges;
+		}
+	}
+
+	for (i = 0; i < iio_aux->num_chans; i++) {
+		iio_aux_chan = iio_aux->chans + i;
+		iio_aux_chan->name = names[i];
+		iio_aux_chan->is_invert_range = invert_ranges[i];
+
+		iio_aux_chan->iio_chan = devm_iio_channel_get(dev, iio_aux_chan->name);
+		if (IS_ERR(iio_aux_chan->iio_chan)) {
+			ret = PTR_ERR(iio_aux_chan->iio_chan);
+			dev_err_probe(dev, ret, "get IIO channel '%s' failed\n",
+				      iio_aux_chan->name);
+			goto out_free_invert_ranges;
+		}
+	}
+
+	platform_set_drvdata(pdev, iio_aux);
+
+	ret = devm_snd_soc_register_component(dev, &audio_iio_aux_component_driver,
+					      NULL, 0);
+out_free_invert_ranges:
+	kfree(invert_ranges);
+out_free_names:
+	kfree(names);
+	return ret;
+}
+
+static const struct of_device_id audio_iio_aux_ids[] = {
+	{ .compatible = "audio-iio-aux" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, audio_iio_aux_ids);
+
+static struct platform_driver audio_iio_aux_driver = {
+	.driver = {
+		.name = "audio-iio-aux",
+		.of_match_table = audio_iio_aux_ids,
+	},
+	.probe = audio_iio_aux_probe,
+};
+module_platform_driver(audio_iio_aux_driver);
+
+MODULE_AUTHOR("Herve Codina <herve.codina@...tlin.com>");
+MODULE_DESCRIPTION("IIO ALSA SoC aux driver");
+MODULE_LICENSE("GPL");
-- 
2.40.1

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