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-rw-r--r--sound/soc/meson/Kconfig9
-rw-r--r--sound/soc/meson/Makefile2
-rw-r--r--sound/soc/meson/axg-card.c87
-rw-r--r--sound/soc/meson/axg-tdm-formatter.c29
-rw-r--r--sound/soc/meson/axg-tdm-interface.c4
-rw-r--r--sound/soc/meson/axg-tdm.h2
-rw-r--r--sound/soc/meson/axg-tdmin.c1
-rw-r--r--sound/soc/meson/axg-tdmout.c1
-rw-r--r--sound/soc/meson/g12a-tohdmitx.c413
9 files changed, 524 insertions, 24 deletions
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 8779fe23671d..63b38c123103 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -1,3 +1,4 @@
+# SPDX-License-Identifier: GPL-2.0-only
menu "ASoC support for Amlogic platforms"
depends on ARCH_MESON || COMPILE_TEST
@@ -56,6 +57,7 @@ config SND_MESON_AXG_SOUND_CARD
imply SND_MESON_AXG_SPDIFOUT
imply SND_MESON_AXG_SPDIFIN
imply SND_MESON_AXG_PDM
+ imply SND_MESON_G12A_TOHDMITX if DRM_MESON_DW_HDMI
help
Select Y or M to add support for the AXG SoC sound card
@@ -82,4 +84,11 @@ config SND_MESON_AXG_PDM
help
Select Y or M to add support for PDM input embedded
in the Amlogic AXG SoC family
+
+config SND_MESON_G12A_TOHDMITX
+ tristate "Amlogic G12A To HDMI TX Control Support"
+ imply SND_SOC_HDMI_CODEC
+ help
+ Select Y or M to add support for HDMI audio on the g12a SoC
+ family
endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
index b45dfb9e2f88..1a8b1470ed84 100644
--- a/sound/soc/meson/Makefile
+++ b/sound/soc/meson/Makefile
@@ -11,6 +11,7 @@ snd-soc-meson-axg-sound-card-objs := axg-card.o
snd-soc-meson-axg-spdifin-objs := axg-spdifin.o
snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
snd-soc-meson-axg-pdm-objs := axg-pdm.o
+snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o
obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
@@ -23,3 +24,4 @@ obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o
obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o
obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o
+obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index aa54d2c612c9..14a8321744da 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -29,6 +29,18 @@ struct axg_dai_link_tdm_data {
struct axg_dai_link_tdm_mask *codec_masks;
};
+/*
+ * Base params for the codec to codec links
+ * Those will be over-written by the CPU side of the link
+ */
+static const struct snd_soc_pcm_stream codec_params = {
+ .formats = SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 5525,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 8,
+};
+
#define PREFIX "amlogic,"
static int axg_card_reallocate_links(struct axg_card *priv,
@@ -80,10 +92,11 @@ static int axg_card_parse_dai(struct snd_soc_card *card,
static int axg_card_set_link_name(struct snd_soc_card *card,
struct snd_soc_dai_link *link,
+ struct device_node *node,
const char *prefix)
{
char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
- prefix, link->cpu_of_node->full_name);
+ prefix, node->full_name);
if (!name)
return -ENOMEM;
@@ -102,7 +115,8 @@ static void axg_card_clean_references(struct axg_card *priv)
if (card->dai_link) {
for_each_card_prelinks(card, i, link) {
- of_node_put(link->cpu_of_node);
+ if (link->cpus)
+ of_node_put(link->cpus->of_node);
for_each_link_codecs(link, j, codec)
of_node_put(codec->of_node);
}
@@ -241,6 +255,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
struct axg_card *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_dai_link *pad = &card->dai_link[*index];
struct snd_soc_dai_link *lb;
+ struct snd_soc_dai_link_component *dlc;
int ret;
/* extend links */
@@ -254,11 +269,20 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
if (!lb->name)
return -ENOMEM;
+ dlc = devm_kzalloc(card->dev, 2 * sizeof(*dlc), GFP_KERNEL);
+ if (!dlc)
+ return -ENOMEM;
+
+ lb->cpus = &dlc[0];
+ lb->codecs = &dlc[1];
+ lb->num_cpus = 1;
+ lb->num_codecs = 1;
+
lb->stream_name = lb->name;
- lb->cpu_of_node = pad->cpu_of_node;
- lb->cpu_dai_name = "TDM Loopback";
- lb->codec_name = "snd-soc-dummy";
- lb->codec_dai_name = "snd-soc-dummy-dai";
+ lb->cpus->of_node = pad->cpus->of_node;
+ lb->cpus->dai_name = "TDM Loopback";
+ lb->codecs->name = "snd-soc-dummy";
+ lb->codecs->dai_name = "snd-soc-dummy-dai";
lb->dpcm_capture = 1;
lb->no_pcm = 1;
lb->ops = &axg_card_tdm_be_ops;
@@ -271,7 +295,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
* axg_card_clean_references() will iterate over this link,
* make sure the node count is balanced
*/
- of_node_get(lb->cpu_of_node);
+ of_node_get(lb->cpus->of_node);
/* Let add_links continue where it should */
*index += 1;
@@ -413,7 +437,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card,
/* Setup tdm link */
link->ops = &axg_card_tdm_be_ops;
link->init = axg_card_tdm_dai_init;
- link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node);
+ link->dai_fmt = axg_card_parse_daifmt(node, link->cpus->of_node);
of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
@@ -474,7 +498,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card,
codec++;
}
- ret = axg_card_set_link_name(card, link, "be");
+ ret = axg_card_set_link_name(card, link, node, "be");
if (ret)
dev_err(card->dev, "error setting %pOFn link name\n", np);
@@ -483,21 +507,31 @@ static int axg_card_set_be_link(struct snd_soc_card *card,
static int axg_card_set_fe_link(struct snd_soc_card *card,
struct snd_soc_dai_link *link,
+ struct device_node *node,
bool is_playback)
{
+ struct snd_soc_dai_link_component *codec;
+
+ codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL);
+ if (!codec)
+ return -ENOMEM;
+
+ link->codecs = codec;
+ link->num_codecs = 1;
+
link->dynamic = 1;
link->dpcm_merged_format = 1;
link->dpcm_merged_chan = 1;
link->dpcm_merged_rate = 1;
- link->codec_dai_name = "snd-soc-dummy-dai";
- link->codec_name = "snd-soc-dummy";
+ link->codecs->dai_name = "snd-soc-dummy-dai";
+ link->codecs->name = "snd-soc-dummy";
if (is_playback)
link->dpcm_playback = 1;
else
link->dpcm_capture = 1;
- return axg_card_set_link_name(card, link, "fe");
+ return axg_card_set_link_name(card, link, node, "fe");
}
static int axg_card_cpu_is_capture_fe(struct device_node *np)
@@ -515,29 +549,44 @@ static int axg_card_cpu_is_tdm_iface(struct device_node *np)
return of_device_is_compatible(np, PREFIX "axg-tdm-iface");
}
+static int axg_card_cpu_is_codec(struct device_node *np)
+{
+ return of_device_is_compatible(np, PREFIX "g12a-tohdmitx");
+}
+
static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
int *index)
{
struct snd_soc_dai_link *dai_link = &card->dai_link[*index];
+ struct snd_soc_dai_link_component *cpu;
int ret;
- ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node,
- &dai_link->cpu_dai_name);
+ cpu = devm_kzalloc(card->dev, sizeof(*cpu), GFP_KERNEL);
+ if (!cpu)
+ return -ENOMEM;
+
+ dai_link->cpus = cpu;
+ dai_link->num_cpus = 1;
+
+ ret = axg_card_parse_dai(card, np, &dai_link->cpus->of_node,
+ &dai_link->cpus->dai_name);
if (ret)
return ret;
- if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node))
- ret = axg_card_set_fe_link(card, dai_link, true);
- else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node))
- ret = axg_card_set_fe_link(card, dai_link, false);
+ if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node))
+ ret = axg_card_set_fe_link(card, dai_link, np, true);
+ else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node))
+ ret = axg_card_set_fe_link(card, dai_link, np, false);
else
ret = axg_card_set_be_link(card, dai_link, np);
if (ret)
return ret;
- if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node))
+ if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
ret = axg_card_parse_tdm(card, np, index);
+ else if (axg_card_cpu_is_codec(dai_link->cpus->of_node))
+ dai_link->params = &codec_params;
return ret;
}
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 0c6cce5c5773..2e498201139f 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -7,6 +7,7 @@
#include <linux/module.h>
#include <linux/of_platform.h>
#include <linux/regmap.h>
+#include <linux/reset.h>
#include <sound/soc.h>
#include "axg-tdm-formatter.h"
@@ -20,6 +21,7 @@ struct axg_tdm_formatter {
struct clk *lrclk;
struct clk *sclk_sel;
struct clk *lrclk_sel;
+ struct reset_control *reset;
bool enabled;
struct regmap *map;
};
@@ -76,6 +78,24 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
return 0;
/*
+ * On the g12a (and possibly other SoCs), when a stream using
+ * multiple lanes is restarted, it will sometimes not start
+ * from the first lane, but randomly from another used one.
+ * The result is an unexpected and random channel shift.
+ *
+ * The hypothesis is that an HW counter is not properly reset
+ * and the formatter simply starts on the lane it stopped
+ * before. Unfortunately, there does not seems to be a way to
+ * reset this through the registers of the block.
+ *
+ * However, the g12a has indenpendent reset lines for each audio
+ * devices. Using this reset before each start solves the issue.
+ */
+ ret = reset_control_reset(formatter->reset);
+ if (ret)
+ return ret;
+
+ /*
* If sclk is inverted, invert it back and provide the inversion
* required by the formatter
*/
@@ -306,6 +326,15 @@ int axg_tdm_formatter_probe(struct platform_device *pdev)
return ret;
}
+ /* Formatter dedicated reset line */
+ formatter->reset = reset_control_get_optional_exclusive(dev, NULL);
+ if (IS_ERR(formatter->reset)) {
+ ret = PTR_ERR(formatter->reset);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to get reset: %d\n", ret);
+ return ret;
+ }
+
return devm_snd_soc_register_component(dev, drv->component_drv,
NULL, 0);
}
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 585ce030b79b..d51f3344be7c 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -306,8 +306,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
}
break;
- case SND_SOC_DAI_FORMAT_DSP_A:
- case SND_SOC_DAI_FORMAT_DSP_B:
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
break;
default:
diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h
index e578b6f40a07..5774ce0916d4 100644
--- a/sound/soc/meson/axg-tdm.h
+++ b/sound/soc/meson/axg-tdm.h
@@ -40,7 +40,7 @@ struct axg_tdm_iface {
static inline bool axg_tdm_lrclk_invert(unsigned int fmt)
{
- return (fmt & SND_SOC_DAIFMT_I2S) ^
+ return ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) ^
!!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF));
}
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index a790f925a4ef..cb87f17f3e95 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -121,7 +121,6 @@ static int axg_tdmin_prepare(struct regmap *map,
break;
case SND_SOC_DAIFMT_LEFT_J:
- case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_DSP_B:
break;
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 527bfc4487e0..86537fc0ecb5 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -137,7 +137,6 @@ static int axg_tdmout_prepare(struct regmap *map,
break;
case SND_SOC_DAIFMT_LEFT_J:
- case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_DSP_B:
skew += 1;
break;
diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c
new file mode 100644
index 000000000000..707ccb192e4c
--- /dev/null
+++ b/sound/soc/meson/g12a-tohdmitx.c
@@ -0,0 +1,413 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2019 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-g12a-tohdmitx.h>
+
+#define G12A_TOHDMITX_DRV_NAME "g12a-tohdmitx"
+
+#define TOHDMITX_CTRL0 0x0
+#define CTRL0_ENABLE_SHIFT 31
+#define CTRL0_I2S_DAT_SEL GENMASK(13, 12)
+#define CTRL0_I2S_LRCLK_SEL GENMASK(9, 8)
+#define CTRL0_I2S_BLK_CAP_INV BIT(7)
+#define CTRL0_I2S_BCLK_O_INV BIT(6)
+#define CTRL0_I2S_BCLK_SEL GENMASK(5, 4)
+#define CTRL0_SPDIF_CLK_CAP_INV BIT(3)
+#define CTRL0_SPDIF_CLK_O_INV BIT(2)
+#define CTRL0_SPDIF_SEL BIT(1)
+#define CTRL0_SPDIF_CLK_SEL BIT(0)
+
+struct g12a_tohdmitx_input {
+ struct snd_pcm_hw_params params;
+ unsigned int fmt;
+};
+
+static struct snd_soc_dapm_widget *
+g12a_tohdmitx_get_input(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_path *p = NULL;
+ struct snd_soc_dapm_widget *in;
+
+ snd_soc_dapm_widget_for_each_source_path(w, p) {
+ if (!p->connect)
+ continue;
+
+ /* Check that we still are in the same component */
+ if (snd_soc_dapm_to_component(w->dapm) !=
+ snd_soc_dapm_to_component(p->source->dapm))
+ continue;
+
+ if (p->source->id == snd_soc_dapm_dai_in)
+ return p->source;
+
+ in = g12a_tohdmitx_get_input(p->source);
+ if (in)
+ return in;
+ }
+
+ return NULL;
+}
+
+static struct g12a_tohdmitx_input *
+g12a_tohdmitx_get_input_data(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_widget *in =
+ g12a_tohdmitx_get_input(w);
+ struct snd_soc_dai *dai;
+
+ if (WARN_ON(!in))
+ return NULL;
+
+ dai = in->priv;
+
+ return dai->playback_dma_data;
+}
+
+static const char * const g12a_tohdmitx_i2s_mux_texts[] = {
+ "I2S A", "I2S B", "I2S C",
+};
+
+static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_i2s_mux_enum,
+ g12a_tohdmitx_i2s_mux_texts);
+
+static int g12a_tohdmitx_get_input_val(struct snd_soc_component *component,
+ unsigned int mask)
+{
+ unsigned int val;
+
+ snd_soc_component_read(component, TOHDMITX_CTRL0, &val);
+ return (val & mask) >> __ffs(mask);
+}
+
+static int g12a_tohdmitx_i2s_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_kcontrol_component(kcontrol);
+
+ ucontrol->value.enumerated.item[0] =
+ g12a_tohdmitx_get_input_val(component, CTRL0_I2S_DAT_SEL);
+
+ return 0;
+}
+
+static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_kcontrol_component(kcontrol);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int mux = ucontrol->value.enumerated.item[0];
+ unsigned int val = g12a_tohdmitx_get_input_val(component,
+ CTRL0_I2S_DAT_SEL);
+
+ /* Force disconnect of the mux while updating */
+ if (val != mux)
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+ snd_soc_component_update_bits(component, TOHDMITX_CTRL0,
+ CTRL0_I2S_DAT_SEL |
+ CTRL0_I2S_LRCLK_SEL |
+ CTRL0_I2S_BCLK_SEL,
+ FIELD_PREP(CTRL0_I2S_DAT_SEL, mux) |
+ FIELD_PREP(CTRL0_I2S_LRCLK_SEL, mux) |
+ FIELD_PREP(CTRL0_I2S_BCLK_SEL, mux));
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux =
+ SOC_DAPM_ENUM_EXT("I2S Source", g12a_tohdmitx_i2s_mux_enum,
+ g12a_tohdmitx_i2s_mux_get_enum,
+ g12a_tohdmitx_i2s_mux_put_enum);
+
+static const char * const g12a_tohdmitx_spdif_mux_texts[] = {
+ "SPDIF A", "SPDIF B",
+};
+
+static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_spdif_mux_enum,
+ g12a_tohdmitx_spdif_mux_texts);
+
+static int g12a_tohdmitx_spdif_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_kcontrol_component(kcontrol);
+
+ ucontrol->value.enumerated.item[0] =
+ g12a_tohdmitx_get_input_val(component, CTRL0_SPDIF_SEL);
+
+ return 0;
+}
+
+static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_kcontrol_component(kcontrol);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int mux = ucontrol->value.enumerated.item[0];
+ unsigned int val = g12a_tohdmitx_get_input_val(component,
+ CTRL0_SPDIF_SEL);
+
+ /* Force disconnect of the mux while updating */
+ if (val != mux)
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+ snd_soc_component_update_bits(component, TOHDMITX_CTRL0,
+ CTRL0_SPDIF_SEL |
+ CTRL0_SPDIF_CLK_SEL,
+ FIELD_PREP(CTRL0_SPDIF_SEL, mux) |
+ FIELD_PREP(CTRL0_SPDIF_CLK_SEL, mux));
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new g12a_tohdmitx_spdif_mux =
+ SOC_DAPM_ENUM_EXT("SPDIF Source", g12a_tohdmitx_spdif_mux_enum,
+ g12a_tohdmitx_spdif_mux_get_enum,
+ g12a_tohdmitx_spdif_mux_put_enum);
+
+static const struct snd_kcontrol_new g12a_tohdmitx_out_enable =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOHDMITX_CTRL0,
+ CTRL0_ENABLE_SHIFT, 1, 0);
+
+static const struct snd_soc_dapm_widget g12a_tohdmitx_widgets[] = {
+ SND_SOC_DAPM_MUX("I2S SRC", SND_SOC_NOPM, 0, 0,
+ &g12a_tohdmitx_i2s_mux),
+ SND_SOC_DAPM_SWITCH("I2S OUT EN", SND_SOC_NOPM, 0, 0,
+ &g12a_tohdmitx_out_enable),
+ SND_SOC_DAPM_MUX("SPDIF SRC", SND_SOC_NOPM, 0, 0,
+ &g12a_tohdmitx_spdif_mux),
+ SND_SOC_DAPM_SWITCH("SPDIF OUT EN", SND_SOC_NOPM, 0, 0,
+ &g12a_tohdmitx_out_enable),
+};
+
+static int g12a_tohdmitx_input_probe(struct snd_soc_dai *dai)
+{
+ struct g12a_tohdmitx_input *data;
+
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ dai->playback_dma_data = data;
+ return 0;
+}
+
+static int g12a_tohdmitx_input_remove(struct snd_soc_dai *dai)
+{
+ kfree(dai->playback_dma_data);
+ return 0;
+}
+
+static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct g12a_tohdmitx_input *data = dai->playback_dma_data;
+
+ /* Save the stream params for the downstream link */
+ memcpy(&data->params, params, sizeof(*params));
+
+ return 0;
+}
+
+static int g12a_tohdmitx_output_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct g12a_tohdmitx_input *in_data =
+ g12a_tohdmitx_get_input_data(dai->capture_widget);
+
+ if (!in_data)
+ return -ENODEV;
+
+ memcpy(params, &in_data->params, sizeof(*params));
+
+ return 0;
+}
+
+static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt)
+{
+ struct g12a_tohdmitx_input *data = dai->playback_dma_data;
+
+ /* Save the source stream format for the downstream link */
+ data->fmt = fmt;
+ return 0;
+}
+
+static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct g12a_tohdmitx_input *in_data =
+ g12a_tohdmitx_get_input_data(dai->capture_widget);
+
+ if (!in_data)
+ return -ENODEV;
+
+ if (!in_data->fmt)
+ return 0;
+
+ return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt);
+}
+
+static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = {
+ .hw_params = g12a_tohdmitx_input_hw_params,
+ .set_fmt = g12a_tohdmitx_input_set_fmt,
+};
+
+static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = {
+ .hw_params = g12a_tohdmitx_output_hw_params,
+ .startup = g12a_tohdmitx_output_startup,
+};
+
+#define TOHDMITX_SPDIF_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+#define TOHDMITX_I2S_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+#define TOHDMITX_STREAM(xname, xsuffix, xfmt, xchmax) \
+{ \
+ .stream_name = xname " " xsuffix, \
+ .channels_min = 1, \
+ .channels_max = (xchmax), \
+ .rate_min = 8000, \
+ .rate_max = 192000, \
+ .formats = (xfmt), \
+}
+
+#define TOHDMITX_IN(xname, xid, xfmt, xchmax) { \
+ .name = xname, \
+ .id = (xid), \
+ .playback = TOHDMITX_STREAM(xname, "Playback", xfmt, xchmax), \
+ .ops = &g12a_tohdmitx_input_ops, \
+ .probe = g12a_tohdmitx_input_probe, \
+ .remove = g12a_tohdmitx_input_remove, \
+}
+
+#define TOHDMITX_OUT(xname, xid, xfmt, xchmax) { \
+ .name = xname, \
+ .id = (xid), \
+ .capture = TOHDMITX_STREAM(xname, "Capture", xfmt, xchmax), \
+ .ops = &g12a_tohdmitx_output_ops, \
+}
+
+static struct snd_soc_dai_driver g12a_tohdmitx_dai_drv[] = {
+ TOHDMITX_IN("I2S IN A", TOHDMITX_I2S_IN_A,
+ TOHDMITX_I2S_FORMATS, 8),
+ TOHDMITX_IN("I2S IN B", TOHDMITX_I2S_IN_B,
+ TOHDMITX_I2S_FORMATS, 8),
+ TOHDMITX_IN("I2S IN C", TOHDMITX_I2S_IN_C,
+ TOHDMITX_I2S_FORMATS, 8),
+ TOHDMITX_OUT("I2S OUT", TOHDMITX_I2S_OUT,
+ TOHDMITX_I2S_FORMATS, 8),
+ TOHDMITX_IN("SPDIF IN A", TOHDMITX_SPDIF_IN_A,
+ TOHDMITX_SPDIF_FORMATS, 2),
+ TOHDMITX_IN("SPDIF IN B", TOHDMITX_SPDIF_IN_B,
+ TOHDMITX_SPDIF_FORMATS, 2),
+ TOHDMITX_OUT("SPDIF OUT", TOHDMITX_SPDIF_OUT,
+ TOHDMITX_SPDIF_FORMATS, 2),
+};
+
+static int g12a_tohdmi_component_probe(struct snd_soc_component *c)
+{
+ /* Initialize the static clock parameters */
+ return snd_soc_component_write(c, TOHDMITX_CTRL0,
+ CTRL0_I2S_BLK_CAP_INV | CTRL0_SPDIF_CLK_CAP_INV);
+}
+
+static const struct snd_soc_dapm_route g12a_tohdmitx_routes[] = {
+ { "I2S SRC", "I2S A", "I2S IN A Playback" },
+ { "I2S SRC", "I2S B", "I2S IN B Playback" },
+ { "I2S SRC", "I2S C", "I2S IN C Playback" },
+ { "I2S OUT EN", "Switch", "I2S SRC" },
+ { "I2S OUT Capture", NULL, "I2S OUT EN" },
+ { "SPDIF SRC", "SPDIF A", "SPDIF IN A Playback" },
+ { "SPDIF SRC", "SPDIF B", "SPDIF IN B Playback" },
+ { "SPDIF OUT EN", "Switch", "SPDIF SRC" },
+ { "SPDIF OUT Capture", NULL, "SPDIF OUT EN" },
+};
+
+static const struct snd_soc_component_driver g12a_tohdmitx_component_drv = {
+ .probe = g12a_tohdmi_component_probe,
+ .dapm_widgets = g12a_tohdmitx_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(g12a_tohdmitx_widgets),
+ .dapm_routes = g12a_tohdmitx_routes,
+ .num_dapm_routes = ARRAY_SIZE(g12a_tohdmitx_routes),
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+static const struct regmap_config g12a_tohdmitx_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+};
+
+static const struct of_device_id g12a_tohdmitx_of_match[] = {
+ { .compatible = "amlogic,g12a-tohdmitx", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, g12a_tohdmitx_of_match);
+
+static int g12a_tohdmitx_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct resource *res;
+ void __iomem *regs;
+ struct regmap *map;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ map = devm_regmap_init_mmio(dev, regs, &g12a_tohdmitx_regmap_cfg);
+ if (IS_ERR(map)) {
+ dev_err(dev, "failed to init regmap: %ld\n",
+ PTR_ERR(map));
+ return PTR_ERR(map);
+ }
+
+ return devm_snd_soc_register_component(dev,
+ &g12a_tohdmitx_component_drv, g12a_tohdmitx_dai_drv,
+ ARRAY_SIZE(g12a_tohdmitx_dai_drv));
+}
+
+static struct platform_driver g12a_tohdmitx_pdrv = {
+ .driver = {
+ .name = G12A_TOHDMITX_DRV_NAME,
+ .of_match_table = g12a_tohdmitx_of_match,
+ },
+ .probe = g12a_tohdmitx_probe,
+};
+module_platform_driver(g12a_tohdmitx_pdrv);
+
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_DESCRIPTION("Amlogic G12a To HDMI Tx Control Codec Driver");
+MODULE_LICENSE("GPL v2");