| Message ID | 20230623085830.749991-13-herve.codina@bootlin.com |
|---|---|
| State | Accepted |
| Commit | 1d298ad822178d365b53eac298c1752730505306 |
| Headers | show |
| Series | Add support for IIO devices in ASoC | expand |
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@bootlin.com> > Reviewed-by: Jonathan Cameron <Jonathan.Cameron@huawei.com> Reviewed-by: Christophe Leroy <christophe.leroy@csgroup.eu> > --- > 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@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; > + 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@bootlin.com>"); > +MODULE_DESCRIPTION("IIO ALSA SoC aux driver"); > +MODULE_LICENSE("GPL");
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@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; + 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@bootlin.com>"); +MODULE_DESCRIPTION("IIO ALSA SoC aux driver"); +MODULE_LICENSE("GPL");