Author | Tokens | Token Proportion | Commits | Commit Proportion |
---|---|---|---|---|
Herve Codina | 1432 | 93.90% | 1 | 25.00% |
Krzysztof Kozlowski | 67 | 4.39% | 2 | 50.00% |
Christophe Jaillet | 26 | 1.70% | 1 | 25.00% |
Total | 1525 | 4 |
// 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/cleanup.h> #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); int ret; /* Allocated names are not needed afterwards (duplicated in ASoC internals) */ char *input_name __free(kfree) = kasprintf(GFP_KERNEL, "%s IN", chan->name); if (!input_name) return -ENOMEM; char *output_name __free(kfree) = kasprintf(GFP_KERNEL, "%s OUT", chan->name); if (!output_name) return -ENOMEM; char *pga_name __free(kfree) = kasprintf(GFP_KERNEL, "%s PGA", chan->name); if (!pga_name) return -ENOMEM; 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) return ret; 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; return snd_soc_dapm_add_routes(dapm, routes, 2); } 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; 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; const char **names __free(kfree) = kcalloc(iio_aux->num_chans, sizeof(*names), GFP_KERNEL); if (!names) return -ENOMEM; u32 *invert_ranges __free(kfree) = kcalloc(iio_aux->num_chans, sizeof(*invert_ranges), GFP_KERNEL); if (!invert_ranges) return -ENOMEM; ret = device_property_read_string_array(dev, "io-channel-names", names, iio_aux->num_chans); if (ret < 0) return dev_err_probe(dev, ret, "failed to read io-channel-names\n"); /* * 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) return dev_err_probe(dev, ret, "failed to read snd-control-invert-range\n"); } 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)) return dev_err_probe(dev, PTR_ERR(iio_aux_chan->iio_chan), "get IIO channel '%s' failed\n", iio_aux_chan->name); } platform_set_drvdata(pdev, iio_aux); return devm_snd_soc_register_component(dev, &audio_iio_aux_component_driver, NULL, 0); } 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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