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// 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@bootlin.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;
unsigned int num_chans;
struct audio_iio_aux_chan chans[] __counted_by(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;
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 = devm_kzalloc(dev, struct_size(iio_aux, chans, count), GFP_KERNEL);
if (!iio_aux)
return -ENOMEM;
iio_aux->dev = dev;
iio_aux->num_chans = count;
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@bootlin.com>");
MODULE_DESCRIPTION("IIO ALSA SoC aux driver");
MODULE_LICENSE("GPL");
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