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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-11 08:27:49 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-11 08:27:49 +0000
commitace9429bb58fd418f0c81d4c2835699bddf6bde6 (patch)
treeb2d64bc10158fdd5497876388cd68142ca374ed3 /sound/soc/codecs/audio-iio-aux.c
parentInitial commit. (diff)
downloadlinux-ace9429bb58fd418f0c81d4c2835699bddf6bde6.tar.xz
linux-ace9429bb58fd418f0c81d4c2835699bddf6bde6.zip
Adding upstream version 6.6.15.upstream/6.6.15
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'sound/soc/codecs/audio-iio-aux.c')
-rw-r--r--sound/soc/codecs/audio-iio-aux.c344
1 files changed, 344 insertions, 0 deletions
diff --git a/sound/soc/codecs/audio-iio-aux.c b/sound/soc/codecs/audio-iio-aux.c
new file mode 100644
index 0000000000..a8bf14239b
--- /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");