Re: [PATCH v6 12/13] ASoC: codecs: Add support for the generic IIO auxiliary devices

[Christophe Leroy]

Le 23/06/2023 à 10:58, Herve Codina a écrit :
> 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@xxxxxxxxxxx>
> Reviewed-by: Jonathan Cameron <Jonathan.Cameron@xxxxxxxxxx>

Reviewed-by: Christophe Leroy <christophe.leroy@xxxxxxxxxx>

> ---
>   sound/soc/codecs/Kconfig         |  12 ++
>   sound/soc/codecs/Makefile        |   2 +
>   sound/soc/codecs/audio-iio-aux.c | 344 +++++++++++++++++++++++++++++++
>   3 files changed, 358 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..a8bf14239bd7
> --- /dev/null
> +++ b/sound/soc/codecs/audio-iio-aux.c
> @@ -0,0 +1,344 @@
> +// 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@xxxxxxxxxxx>
> +
> +#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 = {
> +		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
> +		.name = chan->name,
> +		.info = audio_iio_aux_info_volsw,
> +		.get = audio_iio_aux_get_volsw,
> +		.put = audio_iio_aux_put_volsw,
> +		.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) {
> +			/*
> +			 * This should never happen but to avoid any check
> +			 * later, just swap values here to ensure that the
> +			 * minimum value is lower than the maximum value.
> +			 */
> +			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@xxxxxxxxxxx>");
> +MODULE_DESCRIPTION("IIO ALSA SoC aux driver");
> +MODULE_LICENSE("GPL");