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-rw-r--r--sound/soc/apple/Kconfig16
-rw-r--r--sound/soc/apple/Makefile4
-rw-r--r--sound/soc/apple/macaudio.c1024
-rw-r--r--sound/soc/apple/mca.c23
-rw-r--r--sound/soc/codecs/Kconfig5
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/cirrus,cs42l84.yaml60
-rw-r--r--sound/soc/codecs/cs42l42.c3
-rw-r--r--sound/soc/codecs/cs42l84.c1085
-rw-r--r--sound/soc/codecs/cs42l84.h217
-rw-r--r--sound/soc/codecs/tas2764.c74
-rw-r--r--sound/soc/codecs/tas2764.h7
-rw-r--r--sound/soc/codecs/tas2770.c20
-rw-r--r--sound/soc/codecs/tas2780.c19
-rw-r--r--sound/soc/soc-card.c12
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dapm.c137
-rw-r--r--sound/soc/soc-ops.c232
18 files changed, 2867 insertions, 78 deletions
diff --git a/sound/soc/apple/Kconfig b/sound/soc/apple/Kconfig
index 793f7782e0d7..dce121733091 100644
--- a/sound/soc/apple/Kconfig
+++ b/sound/soc/apple/Kconfig
@@ -6,3 +6,19 @@ config SND_SOC_APPLE_MCA
help
This option enables an ASoC platform driver for MCA peripherals found
on Apple Silicon SoCs.
+
+config SND_SOC_APPLE_MACAUDIO
+ tristate "Sound support for Apple Silicon Macs"
+ depends on ARCH_APPLE || COMPILE_TEST
+ select SND_SOC_APPLE_MCA
+ select SND_SIMPLE_CARD_UTILS
+ select APPLE_ADMAC
+ select COMMON_CLK_APPLE_NCO
+ select SND_SOC_TAS2764
+ select SND_SOC_TAS2770
+ select SND_SOC_CS42L42
+ default ARCH_APPLE
+ help
+ This option enables an ASoC machine-level driver for Apple Silicon Macs
+ and it also enables the required SoC and codec drivers for overall
+ sound support on these machines.
diff --git a/sound/soc/apple/Makefile b/sound/soc/apple/Makefile
index 7a30bf452817..a14b8fc7f349 100644
--- a/sound/soc/apple/Makefile
+++ b/sound/soc/apple/Makefile
@@ -1,3 +1,7 @@
snd-soc-apple-mca-objs := mca.o
obj-$(CONFIG_SND_SOC_APPLE_MCA) += snd-soc-apple-mca.o
+
+snd-soc-macaudio-objs := macaudio.o
+
+obj-$(CONFIG_SND_SOC_APPLE_MACAUDIO) += snd-soc-macaudio.o
diff --git a/sound/soc/apple/macaudio.c b/sound/soc/apple/macaudio.c
new file mode 100644
index 000000000000..b1b74903db7d
--- /dev/null
+++ b/sound/soc/apple/macaudio.c
@@ -0,0 +1,1024 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * ASoC machine driver for Apple Silicon Macs
+ *
+ * Copyright (C) The Asahi Linux Contributors
+ *
+ * Based on sound/soc/qcom/{sc7180.c|common.c}
+ * Copyright (c) 2018, Linaro Limited.
+ * Copyright (c) 2020, The Linux Foundation. All rights reserved.
+ *
+ *
+ * The platform driver has independent frontend and backend DAIs with the
+ * option of routing backends to any of the frontends. The platform
+ * driver configures the routing based on DPCM couplings in ASoC runtime
+ * structures, which in turn are determined from DAPM paths by ASoC. But the
+ * platform driver doesn't supply relevant DAPM paths and leaves that up for
+ * the machine driver to fill in. The filled-in virtual topology can be
+ * anything as long as any backend isn't connected to more than one frontend
+ * at any given time. (The limitation is due to the unsupported case of
+ * reparenting of live BEs.)
+ */
+
+#define DEBUG
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/simple_card_utils.h>
+#include <sound/soc.h>
+#include <sound/soc-jack.h>
+#include <uapi/linux/input-event-codes.h>
+
+#define DRIVER_NAME "snd-soc-macaudio"
+
+/*
+ * CPU side is bit and frame clock provider
+ * I2S has both clocks inverted
+ */
+#define MACAUDIO_DAI_FMT (SND_SOC_DAIFMT_I2S | \
+ SND_SOC_DAIFMT_CBC_CFC | \
+ SND_SOC_DAIFMT_GATED | \
+ SND_SOC_DAIFMT_IB_IF)
+#define MACAUDIO_JACK_MASK (SND_JACK_HEADSET | SND_JACK_HEADPHONE)
+#define MACAUDIO_SLOTWIDTH 32
+
+struct macaudio_snd_data {
+ struct snd_soc_card card;
+ struct snd_soc_jack jack;
+ int jack_plugin_state;
+
+ bool has_speakers;
+ unsigned int max_channels;
+
+ struct macaudio_link_props {
+ /* frontend props */
+ unsigned int bclk_ratio;
+
+ /* backend props */
+ bool is_speakers;
+ bool is_headphones;
+ unsigned int tdm_mask;
+ } *link_props;
+
+ unsigned int speaker_nchans_array[2];
+ struct snd_pcm_hw_constraint_list speaker_nchans_list;
+};
+
+static bool please_blow_up_my_speakers;
+module_param(please_blow_up_my_speakers, bool, 0644);
+MODULE_PARM_DESC(please_blow_up_my_speakers, "Allow unsafe or untested operating configurations");
+
+SND_SOC_DAILINK_DEFS(primary,
+ DAILINK_COMP_ARRAY(COMP_CPU("mca-pcm-0")), // CPU
+ DAILINK_COMP_ARRAY(COMP_DUMMY()), // CODEC
+ DAILINK_COMP_ARRAY(COMP_EMPTY())); // platform (filled at runtime)
+
+SND_SOC_DAILINK_DEFS(secondary,
+ DAILINK_COMP_ARRAY(COMP_CPU("mca-pcm-1")), // CPU
+ DAILINK_COMP_ARRAY(COMP_DUMMY()), // CODEC
+ DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+static struct snd_soc_dai_link macaudio_fe_links[] = {
+ {
+ .name = "Primary",
+ .stream_name = "Primary",
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .dpcm_merged_rate = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_format = 1,
+ .dai_fmt = MACAUDIO_DAI_FMT,
+ SND_SOC_DAILINK_REG(primary),
+ },
+ {
+ .name = "Secondary",
+ .stream_name = "Secondary",
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_merged_rate = 1,
+ .dpcm_merged_chan = 1,
+ .dpcm_merged_format = 1,
+ .dai_fmt = MACAUDIO_DAI_FMT,
+ SND_SOC_DAILINK_REG(secondary),
+ },
+};
+
+static struct macaudio_link_props macaudio_fe_link_props[] = {
+ {
+ /*
+ * Primary FE
+ *
+ * The bclk ratio at 64 for the primary frontend is important
+ * to ensure that the headphones codec's idea of left and right
+ * in a stereo stream over I2S fits in nicely with everyone else's.
+ * (This is until the headphones codec's driver supports
+ * set_tdm_slot.)
+ *
+ * The low bclk ratio precludes transmitting more than two
+ * channels over I2S, but that's okay since there is the secondary
+ * FE for speaker arrays anyway.
+ */
+ .bclk_ratio = 64,
+ },
+ {
+ /*
+ * Secondary FE
+ *
+ * Here we want frames plenty long to be able to drive all
+ * those fancy speaker arrays.
+ */
+ .bclk_ratio = 256,
+ }
+};
+
+static int macaudio_copy_link(struct device *dev, struct snd_soc_dai_link *target,
+ struct snd_soc_dai_link *source)
+{
+ memcpy(target, source, sizeof(struct snd_soc_dai_link));
+
+ target->cpus = devm_kmemdup(dev, target->cpus,
+ sizeof(*target->cpus) * target->num_cpus,
+ GFP_KERNEL);
+ target->codecs = devm_kmemdup(dev, target->codecs,
+ sizeof(*target->codecs) * target->num_codecs,
+ GFP_KERNEL);
+ target->platforms = devm_kmemdup(dev, target->platforms,
+ sizeof(*target->platforms) * target->num_platforms,
+ GFP_KERNEL);
+
+ if (!target->cpus || !target->codecs || !target->platforms)
+ return -ENOMEM;
+
+ return 0;
+}
+
+static int macaudio_parse_of_component(struct device_node *node, int index,
+ struct snd_soc_dai_link_component *comp)
+{
+ struct of_phandle_args args;
+ int ret;
+
+ ret = of_parse_phandle_with_args(node, "sound-dai", "#sound-dai-cells",
+ index, &args);
+ if (ret)
+ return ret;
+ comp->of_node = args.np;
+ return snd_soc_get_dai_name(&args, &comp->dai_name);
+}
+
+/*
+ * Parse one DPCM backend from the devicetree. This means taking one
+ * of the CPU DAIs and combining it with one or more CODEC DAIs.
+ */
+static int macaudio_parse_of_be_dai_link(struct macaudio_snd_data *ma,
+ struct snd_soc_dai_link *link,
+ int be_index, int ncodecs_per_be,
+ struct device_node *cpu,
+ struct device_node *codec)
+{
+ struct snd_soc_dai_link_component *comp;
+ struct device *dev = ma->card.dev;
+ int codec_base = be_index * ncodecs_per_be;
+ int ret, i;
+
+ link->no_pcm = 1;
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
+
+ link->dai_fmt = MACAUDIO_DAI_FMT;
+
+ link->num_codecs = ncodecs_per_be;
+ link->codecs = devm_kcalloc(dev, ncodecs_per_be,
+ sizeof(*comp), GFP_KERNEL);
+ link->num_cpus = 1;
+ link->cpus = devm_kzalloc(dev, sizeof(*comp), GFP_KERNEL);
+
+ if (!link->codecs || !link->cpus)
+ return -ENOMEM;
+
+ link->num_platforms = 0;
+
+ for_each_link_codecs(link, i, comp) {
+ ret = macaudio_parse_of_component(codec, codec_base + i, comp);
+ if (ret)
+ return ret;
+ }
+
+ ret = macaudio_parse_of_component(cpu, be_index, link->cpus);
+ if (ret)
+ return ret;
+
+ link->name = link->cpus[0].dai_name;
+
+ return 0;
+}
+
+static int macaudio_parse_of(struct macaudio_snd_data *ma)
+{
+ struct device_node *codec = NULL;
+ struct device_node *cpu = NULL;
+ struct device_node *np = NULL;
+ struct device_node *platform = NULL;
+ struct snd_soc_dai_link *link = NULL;
+ struct snd_soc_card *card = &ma->card;
+ struct device *dev = card->dev;
+ struct macaudio_link_props *link_props;
+ int ret, num_links, i;
+
+ ret = snd_soc_of_parse_card_name(card, "model");
+ if (ret) {
+ dev_err(dev, "Error parsing card name: %d\n", ret);
+ return ret;
+ }
+
+ /* Populate links, start with the fixed number of FE links */
+ num_links = ARRAY_SIZE(macaudio_fe_links);
+
+ /* Now add together the (dynamic) number of BE links */
+ for_each_available_child_of_node(dev->of_node, np) {
+ int num_cpus;
+
+ cpu = of_get_child_by_name(np, "cpu");
+ if (!cpu) {
+ dev_err(dev, "missing CPU DAI node at %pOF\n", np);
+ ret = -EINVAL;
+ goto err_free;
+ }
+
+ num_cpus = of_count_phandle_with_args(cpu, "sound-dai",
+ "#sound-dai-cells");
+
+ if (num_cpus <= 0) {
+ dev_err(card->dev, "missing sound-dai property at %pOF\n", cpu);
+ ret = -EINVAL;
+ goto err_free;
+ }
+ of_node_put(cpu);
+ cpu = NULL;
+
+ /* Each CPU specified counts as one BE link */
+ num_links += num_cpus;
+ }
+
+ /* Allocate the DAI link array */
+ card->dai_link = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL);
+ ma->link_props = devm_kcalloc(dev, num_links, sizeof(*ma->link_props), GFP_KERNEL);
+ if (!card->dai_link || !ma->link_props)
+ return -ENOMEM;
+
+ card->num_links = num_links;
+ link = card->dai_link;
+ link_props = ma->link_props;
+
+ for (i = 0; i < ARRAY_SIZE(macaudio_fe_links); i++) {
+ ret = macaudio_copy_link(dev, link, &macaudio_fe_links[i]);
+ if (ret)
+ goto err_free;
+
+ memcpy(link_props, &macaudio_fe_link_props[i], sizeof(struct macaudio_link_props));
+ link++; link_props++;
+ }
+
+ for (i = 0; i < num_links; i++)
+ card->dai_link[i].id = i;
+
+ /* Fill in the BEs */
+ for_each_available_child_of_node(dev->of_node, np) {
+ const char *link_name;
+ bool speakers;
+ int be_index, num_codecs, num_bes, ncodecs_per_cpu, nchannels;
+ unsigned int left_mask, right_mask;
+
+ ret = of_property_read_string(np, "link-name", &link_name);
+ if (ret) {
+ dev_err(card->dev, "missing link name\n");
+ goto err_free;
+ }
+
+ speakers = !strcmp(link_name, "Speaker")
+ || !strcmp(link_name, "Speakers");
+ if (speakers)
+ ma->has_speakers = 1;
+
+ cpu = of_get_child_by_name(np, "cpu");
+ codec = of_get_child_by_name(np, "codec");
+
+ if (!codec || !cpu) {
+ dev_err(dev, "missing DAI specifications for '%s'\n", link_name);
+ ret = -EINVAL;
+ goto err_free;
+ }
+
+ num_bes = of_count_phandle_with_args(cpu, "sound-dai",
+ "#sound-dai-cells");
+ if (num_bes <= 0) {
+ dev_err(card->dev, "missing sound-dai property at %pOF\n", cpu);
+ ret = -EINVAL;
+ goto err_free;
+ }
+
+ num_codecs = of_count_phandle_with_args(codec, "sound-dai",
+ "#sound-dai-cells");
+ if (num_codecs <= 0) {
+ dev_err(card->dev, "missing sound-dai property at %pOF\n", codec);
+ ret = -EINVAL;
+ goto err_free;
+ }
+
+ if (num_codecs % num_bes != 0) {
+ dev_err(card->dev, "bad combination of CODEC (%d) and CPU (%d) number at %pOF\n",
+ num_codecs, num_bes, np);
+ ret = -EINVAL;
+ goto err_free;
+ }
+
+ /*
+ * Now parse the cpu/codec lists into a number of DPCM backend links.
+ * In each link there will be one DAI from the cpu list paired with
+ * an evenly distributed number of DAIs from the codec list. (As is
+ * the binding semantics.)
+ */
+ ncodecs_per_cpu = num_codecs / num_bes;
+ nchannels = num_codecs * (speakers ? 1 : 2);
+
+ /* Save the max number of channels on the platform */
+ if (nchannels > ma->max_channels)
+ ma->max_channels = nchannels;
+
+ /*
+ * If there is a single speaker, assign two channels to it, because
+ * it can do downmix.
+ */
+ if (nchannels < 2)
+ nchannels = 2;
+
+ left_mask = 0;
+ for (i = 0; i < nchannels; i += 2)
+ left_mask = left_mask << 2 | 1;
+ right_mask = left_mask << 1;
+
+ for (be_index = 0; be_index < num_bes; be_index++) {
+ ret = macaudio_parse_of_be_dai_link(ma, link, be_index,
+ ncodecs_per_cpu, cpu, codec);
+ if (ret)
+ goto err_free;
+
+ link_props->is_speakers = speakers;
+ link_props->is_headphones = !speakers;
+
+ if (num_bes == 2)
+ /* This sound peripheral is split between left and right BE */
+ link_props->tdm_mask = be_index ? right_mask : left_mask;
+ else
+ /* One BE covers all of the peripheral */
+ link_props->tdm_mask = left_mask | right_mask;
+
+ /* Steal platform OF reference for use in FE links later */
+ platform = link->cpus->of_node;
+
+ link++; link_props++;
+ }
+
+ of_node_put(codec);
+ of_node_put(cpu);
+ cpu = codec = NULL;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(macaudio_fe_links); i++)
+ card->dai_link[i].platforms->of_node = platform;
+
+ return 0;
+
+err_free:
+ of_node_put(codec);
+ of_node_put(cpu);
+ of_node_put(np);
+
+ if (!card->dai_link)
+ return ret;
+
+ for (i = 0; i < num_links; i++) {
+ /*
+ * TODO: If we don't go through this path are the references
+ * freed inside ASoC?
+ */
+ snd_soc_of_put_dai_link_codecs(&card->dai_link[i]);
+ snd_soc_of_put_dai_link_cpus(&card->dai_link[i]);
+ }
+
+ return ret;
+}
+
+static int macaudio_get_runtime_bclk_ratio(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dpcm *dpcm;
+
+ /*
+ * If this is a FE, look it up in link_props directly.
+ * If this is a BE, look it up in the respective FE.
+ */
+ if (!rtd->dai_link->no_pcm)
+ return ma->link_props[rtd->dai_link->id].bclk_ratio;
+
+ for_each_dpcm_fe(rtd, substream->stream, dpcm) {
+ int fe_id = dpcm->fe->dai_link->id;
+
+ return ma->link_props[fe_id].bclk_ratio;
+ }
+
+ return 0;
+}
+
+static int macaudio_dpcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ int bclk_ratio = macaudio_get_runtime_bclk_ratio(substream);
+ int i;
+
+ if (bclk_ratio) {
+ struct snd_soc_dai *dai;
+ int mclk = params_rate(params) * bclk_ratio;
+
+ for_each_rtd_codec_dais(rtd, i, dai) {
+ snd_soc_dai_set_sysclk(dai, 0, mclk, SND_SOC_CLOCK_IN);
+ snd_soc_dai_set_bclk_ratio(dai, bclk_ratio);
+ }
+
+ snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT);
+ snd_soc_dai_set_bclk_ratio(cpu_dai, bclk_ratio);
+ }
+
+ return 0;
+}
+
+static int macaudio_fe_startup(struct snd_pcm_substream *substream)
+{
+
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(rtd->card);
+ int ret;
+
+ /* The FEs must never have more channels than the hardware */
+ ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 0, ma->max_channels);
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "Failed to constrain FE %d! %d", rtd->dai_link->id, ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int macaudio_fe_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_pcm_runtime *be;
+ struct snd_soc_dpcm *dpcm;
+
+ be = NULL;
+ for_each_dpcm_be(rtd, substream->stream, dpcm) {
+ be = dpcm->be;
+ break;
+ }
+
+ if (!be) {
+ dev_err(rtd->dev, "opening PCM device '%s' with no audio route configured (bad settings applied to the sound card)\n",
+ rtd->dai_link->name);
+ return -EINVAL;
+ }
+
+ return macaudio_dpcm_hw_params(substream, params);
+}
+
+
+static void macaudio_dpcm_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *dai;
+ int bclk_ratio = macaudio_get_runtime_bclk_ratio(substream);
+ int i;
+
+ if (bclk_ratio) {
+ for_each_rtd_codec_dais(rtd, i, dai)
+ snd_soc_dai_set_sysclk(dai, 0, 0, SND_SOC_CLOCK_IN);
+
+ snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT);
+ }
+}
+
+static const struct snd_soc_ops macaudio_fe_ops = {
+ .startup = macaudio_fe_startup,
+ .shutdown = macaudio_dpcm_shutdown,
+ .hw_params = macaudio_fe_hw_params,
+};
+
+static const struct snd_soc_ops macaudio_be_ops = {
+ .shutdown = macaudio_dpcm_shutdown,
+ .hw_params = macaudio_dpcm_hw_params,
+};
+
+static int macaudio_be_assign_tdm(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+ struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id];
+ struct snd_soc_dai *dai;
+ unsigned int mask;
+ int nslots, ret, i;
+
+ if (!props->tdm_mask)
+ return 0;
+
+ mask = props->tdm_mask;
+ nslots = __fls(mask) + 1;
+
+ if (rtd->dai_link->num_codecs == 1) {
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), mask,
+ 0, nslots, MACAUDIO_SLOTWIDTH);
+
+ /*
+ * Headphones get a pass on -ENOTSUPP (see the comment
+ * around bclk_ratio value for primary FE).
+ */
+ if (ret == -ENOTSUPP && props->is_headphones)
+ return 0;
+
+ return ret;
+ }
+
+ for_each_rtd_codec_dais(rtd, i, dai) {
+ int slot = __ffs(mask);
+
+ mask &= ~(1 << slot);
+ ret = snd_soc_dai_set_tdm_slot(dai, 1 << slot, 0, nslots,
+ MACAUDIO_SLOTWIDTH);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int macaudio_be_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+ struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id];
+ struct snd_soc_dai *dai;
+ int i, ret;
+
+ ret = macaudio_be_assign_tdm(rtd);
+ if (ret < 0)
+ return ret;
+
+ if (props->is_headphones) {
+ for_each_rtd_codec_dais(rtd, i, dai)
+ snd_soc_component_set_jack(dai->component, &ma->jack, NULL);
+ }
+
+ return 0;
+}
+
+static void macaudio_be_exit(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+ struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id];
+ struct snd_soc_dai *dai;
+ int i;
+
+ if (props->is_headphones) {
+ for_each_rtd_codec_dais(rtd, i, dai)
+ snd_soc_component_set_jack(dai->component, NULL, NULL);
+ }
+}
+
+static int macaudio_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+ struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id];
+ int nslots = props->bclk_ratio / MACAUDIO_SLOTWIDTH;
+
+ return snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), (1 << nslots) - 1,
+ (1 << nslots) - 1, nslots, MACAUDIO_SLOTWIDTH);
+}
+
+static struct snd_soc_jack_pin macaudio_jack_pins[] = {
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static int macaudio_probe(struct snd_soc_card *card)
+{
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+ int ret;
+
+ dev_dbg(card->dev, "%s!\n", __func__);
+
+ ret = snd_soc_card_jack_new_pins(card, "Headphone Jack",
+ SND_JACK_HEADSET | SND_JACK_HEADPHONE,
+ &ma->jack, macaudio_jack_pins,
+ ARRAY_SIZE(macaudio_jack_pins));
+ if (ret < 0) {
+ dev_err(card->dev, "jack creation failed: %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+static int macaudio_add_backend_dai_route(struct snd_soc_card *card, struct snd_soc_dai *dai,
+ bool is_speakers)
+{
+ struct snd_soc_dapm_route routes[2];
+ struct snd_soc_dapm_route *r;
+ int nroutes = 0;
+ int ret;
+
+ memset(routes, 0, sizeof(routes));
+
+ dev_dbg(card->dev, "adding routes for '%s'\n", dai->name);
+
+ r = &routes[nroutes++];
+ if (is_speakers)
+ r->source = "Speaker Playback";
+ else
+ r->source = "Headphone Playback";
+ r->sink = dai->playback_widget->name;
+
+ /* If headphone jack, add capture path */
+ if (!is_speakers) {
+ r = &routes[nroutes++];
+ r->source = dai->capture_widget->name;
+ r->sink = "Headset Capture";
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, routes, nroutes);
+ if (ret)
+ dev_err(card->dev, "failed adding dynamic DAPM routes for %s\n",
+ dai->name);
+ return ret;
+}
+
+static int macaudio_add_pin_routes(struct snd_soc_card *card, struct snd_soc_component *component,
+ bool is_speakers)
+{
+ struct snd_soc_dapm_route routes[2];
+ struct snd_soc_dapm_route *r;
+ int nroutes = 0;
+ char buf[32];
+ int ret;
+
+ memset(routes, 0, sizeof(routes));
+
+ /* Connect the far ends of CODECs to pins */
+ if (is_speakers) {
+ r = &routes[nroutes++];
+ r->source = "OUT";
+ if (component->name_prefix) {
+ snprintf(buf, sizeof(buf) - 1, "%s OUT", component->name_prefix);
+ r->source = buf;
+ }
+ r->sink = "Speaker";
+ } else {
+ r = &routes[nroutes++];
+ r->source = "Jack HP";
+ r->sink = "Headphone";
+ r = &routes[nroutes++];
+ r->source = "Headset Mic";
+ r->sink = "Jack HS";
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, routes, nroutes);
+ if (ret)
+ dev_err(card->dev, "failed adding dynamic DAPM routes for %s\n",
+ component->name);
+ return ret;
+}
+
+static int macaudio_late_probe(struct snd_soc_card *card)
+{
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+ struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dai *dai;
+ int ret, i;
+
+ /* Add the dynamic DAPM routes */
+ for_each_card_rtds(card, rtd) {
+ struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id];
+
+ if (!rtd->dai_link->no_pcm)
+ continue;
+
+ for_each_rtd_cpu_dais(rtd, i, dai) {
+ ret = macaudio_add_backend_dai_route(card, dai, props->is_speakers);
+
+ if (ret)
+ return ret;
+ }
+
+ for_each_rtd_codec_dais(rtd, i, dai) {
+ ret = macaudio_add_pin_routes(card, dai->component,
+ props->is_speakers);
+
+ if (ret)
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+#define CHECK(call, pattern, value) \
+ { \
+ int ret = call(card, pattern, value); \
+ if (ret < 1 && !please_blow_up_my_speakers) { \
+ dev_err(card->dev, "%s on '%s': %d\n", #call, pattern, ret); \
+ return ret; \
+ } \
+ dev_dbg(card->dev, "%s on '%s': %d hits\n", #call, pattern, ret); \
+ }
+
+
+static int macaudio_j274_fixup_controls(struct snd_soc_card *card)
+{
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+
+ if (ma->has_speakers) {
+ CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 14); // 20 set by macOS, this is 3 dB below
+ }
+
+ return 0;
+}
+
+static int macaudio_j313_fixup_controls(struct snd_soc_card *card) {
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+
+ if (ma->has_speakers) {
+ if (!please_blow_up_my_speakers) {
+ dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n");
+ return -EINVAL;
+ }
+
+ CHECK(snd_soc_set_enum_kctl, "* ASI1 Sel", "Left");
+ CHECK(snd_soc_deactivate_kctl, "* ASI1 Sel", 0);
+
+ /* !!! This is copied from j274, not obtained by looking at
+ * what macOS sets.
+ */
+ CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 14);
+
+ /*
+ * Since we don't set the right slots yet to avoid
+ * driver conflict on the I2S bus sending ISENSE/VSENSE
+ * samples from the codecs back to us, disable the
+ * controls.
+ */
+ CHECK(snd_soc_deactivate_kctl, "* VSENSE Switch", 0);
+ CHECK(snd_soc_deactivate_kctl, "* ISENSE Switch", 0);
+ }
+
+ return 0;
+}
+
+static int macaudio_j314_fixup_controls(struct snd_soc_card *card)
+{
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+
+ if (ma->has_speakers) {
+ if (!please_blow_up_my_speakers) {
+ dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n");
+ return -EINVAL;
+ }
+
+ CHECK(snd_soc_set_enum_kctl, "* ASI1 Sel", "Left");
+ CHECK(snd_soc_deactivate_kctl, "* ASI1 Sel", 0);
+ CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 9); // 15 set by macOS, this is 3 dB below
+ CHECK(snd_soc_set_enum_kctl, "* Tweeter HPF Corner Frequency", "800 Hz");
+ CHECK(snd_soc_deactivate_kctl, "* Tweeter HPF Corner Frequency", 0);
+
+ /*
+ * The speaker amps suffer from spurious overcurrent
+ * events on their unmute, so enable autoretry.
+ */
+ CHECK(snd_soc_set_enum_kctl, "* OCE Handling", "Retry");
+ CHECK(snd_soc_deactivate_kctl, "* OCE Handling", 0);
+
+ /*
+ * Since we don't set the right slots yet to avoid
+ * driver conflict on the I2S bus sending ISENSE/VSENSE
+ * samples from the codecs back to us, disable the
+ * controls.
+ */
+ CHECK(snd_soc_deactivate_kctl, "* VSENSE Switch", 0);
+ CHECK(snd_soc_deactivate_kctl, "* ISENSE Switch", 0);
+ }
+
+ return 0;
+}
+
+static int macaudio_j375_fixup_controls(struct snd_soc_card *card)
+{
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+
+ if (ma->has_speakers) {
+ if (!please_blow_up_my_speakers) {
+ dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n");
+ return -EINVAL;
+ }
+
+ CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 14); // 20 set by macOS, this is 3 dB below
+ }
+
+ return 0;
+}
+
+static int macaudio_j493_fixup_controls(struct snd_soc_card *card)
+{
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+
+ if (ma->has_speakers) {
+ if (!please_blow_up_my_speakers) {
+ dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n");
+ return -EINVAL;
+ }
+
+ CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 9); // 15 set by macOS, this is 3 dB below
+ }
+
+ return 0;
+}
+
+static int macaudio_fallback_fixup_controls(struct snd_soc_card *card)
+{
+ struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
+
+ if (ma->has_speakers && !please_blow_up_my_speakers) {
+ dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+#undef CHECK
+
+static const char * const macaudio_spk_mux_texts[] = {
+ "Primary",
+ "Secondary"
+};
+
+SOC_ENUM_SINGLE_VIRT_DECL(macaudio_spk_mux_enum, macaudio_spk_mux_texts);
+
+static const struct snd_kcontrol_new macaudio_spk_mux =
+ SOC_DAPM_ENUM("Speaker Playback Mux", macaudio_spk_mux_enum);
+
+static const char * const macaudio_hp_mux_texts[] = {
+ "Primary",
+ "Secondary"
+};
+
+SOC_ENUM_SINGLE_VIRT_DECL(macaudio_hp_mux_enum, macaudio_hp_mux_texts);
+
+static const struct snd_kcontrol_new macaudio_hp_mux =
+ SOC_DAPM_ENUM("Headphones Playback Mux", macaudio_hp_mux_enum);
+
+static const struct snd_soc_dapm_widget macaudio_snd_widgets[] = {
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_SPK("Speaker (Static)", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
+ SND_SOC_DAPM_MUX("Speaker Playback Mux", SND_SOC_NOPM, 0, 0, &macaudio_spk_mux),
+ SND_SOC_DAPM_MUX("Headphone Playback Mux", SND_SOC_NOPM, 0, 0, &macaudio_hp_mux),
+
+ SND_SOC_DAPM_AIF_OUT("Speaker Playback", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("Headphone Playback", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_AIF_IN("Headset Capture", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_kcontrol_new macaudio_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static const struct snd_soc_dapm_route macaudio_dapm_routes[] = {
+ /* Playback paths */
+ { "Speaker Playback Mux", "Primary", "PCM0 TX" },
+ { "Speaker Playback Mux", "Secondary", "PCM1 TX" },
+ { "Speaker Playback", NULL, "Speaker Playback Mux"},
+
+ { "Headphone Playback Mux", "Primary", "PCM0 TX" },
+ { "Headphone Playback Mux", "Secondary", "PCM1 TX" },
+ { "Headphone Playback", NULL, "Headphone Playback Mux"},
+ /*
+ * Additional paths (to specific I2S ports) are added dynamically.
+ */
+
+ /* Capture paths */
+ { "PCM0 RX", NULL, "Headset Capture" },
+};
+
+static const struct of_device_id macaudio_snd_device_id[] = {
+ { .compatible = "apple,j274-macaudio", .data = macaudio_j274_fixup_controls },
+ { .compatible = "apple,j313-macaudio", .data = macaudio_j313_fixup_controls },
+ { .compatible = "apple,j314-macaudio", .data = macaudio_j314_fixup_controls },
+ { .compatible = "apple,j375-macaudio", .data = macaudio_j375_fixup_controls },
+ { .compatible = "apple,j413-macaudio", .data = macaudio_j314_fixup_controls },
+ { .compatible = "apple,j493-macaudio", .data = macaudio_j493_fixup_controls },
+ { .compatible = "apple,macaudio"},
+ { }
+};
+MODULE_DEVICE_TABLE(of, macaudio_snd_device_id);
+
+static int macaudio_snd_platform_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card;
+ struct macaudio_snd_data *data;
+ struct device *dev = &pdev->dev;
+ struct snd_soc_dai_link *link;
+ const struct of_device_id *of_id;
+ int ret;
+ int i;
+
+ of_id = of_match_device(macaudio_snd_device_id, dev);
+ if (!of_id)
+ return -EINVAL;
+
+ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+ card = &data->card;
+ snd_soc_card_set_drvdata(card, data);
+
+ card->owner = THIS_MODULE;
+ card->driver_name = "macaudio";
+ card->dev = dev;
+ card->dapm_widgets = macaudio_snd_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(macaudio_snd_widgets);
+ card->dapm_routes = macaudio_dapm_routes;
+ card->num_dapm_routes = ARRAY_SIZE(macaudio_dapm_routes);
+ card->controls = macaudio_controls;
+ card->num_controls = ARRAY_SIZE(macaudio_controls);
+ card->probe = macaudio_probe;
+ card->late_probe = macaudio_late_probe;
+ card->component_chaining = true;
+ card->fully_routed = true;
+
+ if (of_id->data)
+ card->fixup_controls = of_id->data;
+ else
+ card->fixup_controls = macaudio_fallback_fixup_controls;
+
+ ret = macaudio_parse_of(data);
+ if (ret)
+ return dev_err_probe(&pdev->dev, ret, "failed OF parsing\n");
+
+ for_each_card_prelinks(card, i, link) {
+ if (link->no_pcm) {
+ link->ops = &macaudio_be_ops;
+ link->init = macaudio_be_init;
+ link->exit = macaudio_be_exit;
+ } else {
+ link->ops = &macaudio_fe_ops;
+ link->init = macaudio_fe_init;
+ }
+ }
+
+ return devm_snd_soc_register_card(dev, card);
+}
+
+static struct platform_driver macaudio_snd_driver = {
+ .probe = macaudio_snd_platform_probe,
+ .driver = {
+ .name = DRIVER_NAME,
+ .of_match_table = macaudio_snd_device_id,
+ .pm = &snd_soc_pm_ops,
+ },
+};
+module_platform_driver(macaudio_snd_driver);
+
+MODULE_AUTHOR("Martin PoviĊĦer <povik+lin@cutebit.org>");
+MODULE_DESCRIPTION("Apple Silicon Macs machine-level sound driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c
index 24381c42eb54..2b0a3760d3f7 100644
--- a/sound/soc/apple/mca.c
+++ b/sound/soc/apple/mca.c
@@ -456,6 +456,28 @@ err:
return -EINVAL;
}
+static int mca_fe_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct mca_cluster *cl = mca_dai_to_cluster(dai);
+ unsigned int mask, nchannels;
+
+ if (cl->tdm_slots) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mask = cl->tdm_tx_mask;
+ else
+ mask = cl->tdm_rx_mask;
+
+ nchannels = hweight32(mask);
+ } else {
+ nchannels = 2;
+ }
+
+ return snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ 1, nchannels);
+}
+
static int mca_fe_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
unsigned int rx_mask, int slots, int slot_width)
{
@@ -672,6 +694,7 @@ static int mca_fe_hw_params(struct snd_pcm_substream *substream,
}
static const struct snd_soc_dai_ops mca_fe_ops = {
+ .startup = mca_fe_startup,
.set_fmt = mca_fe_set_fmt,
.set_bclk_ratio = mca_set_bclk_ratio,
.set_tdm_slot = mca_fe_set_tdm_slot,
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7022e6286e6c..83ff93f020d2 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS
imply SND_SOC_CS42L52
imply SND_SOC_CS42L56
imply SND_SOC_CS42L73
+ imply SND_SOC_CS42L84
imply SND_SOC_CS4234
imply SND_SOC_CS4265
imply SND_SOC_CS4270
@@ -729,6 +730,10 @@ config SND_SOC_CS42L83
select REGMAP_I2C
select SND_SOC_CS42L42_CORE
+config SND_SOC_CS42L84
+ tristate "Cirrus Logic CS42L84 CODEC"
+ depends on I2C
+
config SND_SOC_CS4234
tristate "Cirrus Logic CS4234 CODEC"
depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9170ee1447dd..2026bb932f47 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -72,6 +72,7 @@ snd-soc-cs42l52-objs := cs42l52.o
snd-soc-cs42l56-objs := cs42l56.o
snd-soc-cs42l73-objs := cs42l73.o
snd-soc-cs42l83-i2c-objs := cs42l83-i2c.o
+snd-soc-cs42l84-objs := cs42l84.o
snd-soc-cs4234-objs := cs4234.o
snd-soc-cs4265-objs := cs4265.o
snd-soc-cs4270-objs := cs4270.o
@@ -432,6 +433,7 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o
obj-$(CONFIG_SND_SOC_CS42L56) += snd-soc-cs42l56.o
obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o
obj-$(CONFIG_SND_SOC_CS42L83) += snd-soc-cs42l83-i2c.o
+obj-$(CONFIG_SND_SOC_CS42L84) += snd-soc-cs42l84.o
obj-$(CONFIG_SND_SOC_CS4234) += snd-soc-cs4234.o
obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
diff --git a/sound/soc/codecs/cirrus,cs42l84.yaml b/sound/soc/codecs/cirrus,cs42l84.yaml
new file mode 100644
index 000000000000..12bc6dbeeddf
--- /dev/null
+++ b/sound/soc/codecs/cirrus,cs42l84.yaml
@@ -0,0 +1,60 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/cirrus,cs42l84.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Cirrus Logic CS42L84 audio CODEC
+
+maintainers:
+ - povik+lin@cutebit.org
+
+description:
+ The CS42L84 is a headphone jack codec made by Cirrus Logic and embedded
+ in personal computers sold by Apple. It was first seen in 2021 Macbook Pro
+ models.
+
+ It has stereo DAC for playback, mono ADC for capture, and is somewhat
+ similar to CS42L42 but with a different regmap.
+
+properties:
+ compatible:
+ enum:
+ - cirrus,cs42l84
+
+ reg:
+ description:
+ I2C address of the device
+ maxItems: 1
+
+ reset-gpios:
+ description:
+ Reset pin, asserted to reset the device, deasserted to bring
+ the device online
+ maxItems: 1
+
+ interrupts:
+ description:
+ Interrupt for the IRQ output line of the device
+ maxItems: 1
+
+required:
+ - compatible
+ - reg
+
+additionalProperties: false
+
+examples:
+ - |
+ i2c {
+ #address-cells = <1>;
+ #size-cells = <0>;
+
+ jack_codec: codec@4b {
+ compatible = "cirrus,cs42l84";
+ reg = <0x4b>;
+ reset-gpios = <&pinctrl_nub 4 GPIO_ACTIVE_LOW>;
+ interrupts-extended = <&pinctrl_ap 180 IRQ_TYPE_LEVEL_LOW>;
+ #sound-dai-cells = <0>;
+ };
+ };
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 2fefbcf7bd13..8d0866cee850 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -1122,7 +1122,6 @@ struct snd_soc_dai_driver cs42l42_dai = {
.formats = CS42L42_FORMATS,
},
.symmetric_rate = 1,
- .symmetric_sample_bits = 1,
.ops = &cs42l42_ops,
};
EXPORT_SYMBOL_NS_GPL(cs42l42_dai, SND_SOC_CS42L42_CORE);
@@ -1648,7 +1647,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data)
return IRQ_NONE;
}
- /* Read sticky registers to clear interurpt */
+ /* Read sticky registers to clear interrupt */
for (i = 0; i < ARRAY_SIZE(stickies); i++) {
regmap_read(cs42l42->regmap, irq_params_table[i].status_addr,
&(stickies[i]));
diff --git a/sound/soc/codecs/cs42l84.c b/sound/soc/codecs/cs42l84.c
new file mode 100644
index 000000000000..47487a1e0f7e
--- /dev/null
+++ b/sound/soc/codecs/cs42l84.c
@@ -0,0 +1,1085 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * cs42l84.c -- CS42L84 ALSA SoC audio driver
+ *
+ * Copyright (C) The Asahi Linux Contributors
+ *
+ * Based on sound/soc/codecs/cs42l42{.c,.h}
+ * Copyright 2016 Cirrus Logic, Inc.
+ */
+
+#define DEBUG
+
+#include <linux/bits.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/acpi.h>
+#include <linux/platform_device.h>
+#include <linux/property.h>
+#include <linux/regulator/consumer.h>
+#include <linux/gpio/consumer.h>
+#include <linux/of_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "cs42l84.h"
+#include "cirrus_legacy.h"
+
+struct cs42l84_private {
+ struct regmap *regmap;
+ struct device *dev;
+ struct gpio_desc *reset_gpio;
+ struct snd_soc_jack *jack;
+ struct mutex irq_lock;
+ u8 plug_state;
+ int pll_config;
+ int bclk;
+ u8 pll_mclk_f;
+ u32 srate;
+ u8 stream_use;
+ int hs_type;
+};
+
+static const struct reg_default cs42l84_reg_defaults[] = {
+};
+
+bool cs42l84_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS42L84_DEVID ... CS42L84_DEVID+5:
+ case CS42L84_TSRS_PLUG_INT_STATUS:
+ case CS42L84_PLL_LOCK_STATUS:
+ case CS42L84_TSRS_PLUG_STATUS:
+ case CS42L84_HS_DET_STATUS2:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config cs42l84_regmap = {
+ .reg_bits = 16,
+ .val_bits = 8,
+
+ .volatile_reg = cs42l84_volatile_register,
+
+ .max_register = 0xffff,
+ /*
+ .reg_defaults = cs42l84_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(cs42l84_reg_defaults),
+ */
+ .cache_type = REGCACHE_RBTREE,
+
+ .use_single_read = true,
+ .use_single_write = true,
+};
+
+static int cs42l84_put_dac_vol(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *val)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kctl);
+ struct soc_mixer_control *mc = (struct soc_mixer_control *) kctl->private_value;
+ int vola, volb;
+ int ret, ret2, updated = 0;
+
+ vola = val->value.integer.value[0] + mc->min;
+ volb = val->value.integer.value[1] + mc->min;
+
+ if (vola < mc->min || vola > mc->max || volb < mc->min || volb > mc->max)
+ return -EINVAL;
+
+ ret = snd_soc_component_update_bits(component, CS42L84_FRZ_CTL,
+ CS42L84_FRZ_CTL_ENGAGE,
+ CS42L84_FRZ_CTL_ENGAGE);
+ if (ret < 0)
+ goto bail;
+ updated |= ret;
+
+ ret = snd_soc_component_update_bits(component, CS42L84_DAC_CHA_VOL_LSB,
+ 0xff, vola & 0xff);
+ if (ret < 0)
+ goto bail;
+ updated |= ret;
+
+ ret = snd_soc_component_update_bits(component, CS42L84_DAC_CHA_VOL_MSB,
+ 0xff, (vola >> 8) & 0x01);
+ if (ret < 0)
+ goto bail;
+ updated |= ret;
+
+ ret = snd_soc_component_update_bits(component, CS42L84_DAC_CHB_VOL_LSB,
+ 0xff, volb & 0xff);
+ if (ret < 0)
+ goto bail;
+ updated |= ret;
+
+ ret = snd_soc_component_update_bits(component, CS42L84_DAC_CHB_VOL_MSB,
+ 0xff, (volb >> 8) & 0x01);
+ if (ret < 0)
+ goto bail;
+ ret |= updated;
+
+bail:
+ ret2 = snd_soc_component_update_bits(component, CS42L84_FRZ_CTL,
+ CS42L84_FRZ_CTL_ENGAGE, 0);
+ if (ret2 < 0 && ret >= 0)
+ ret = ret2;
+
+ return ret;
+}
+
+static int cs42l84_get_dac_vol(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *val)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kctl);
+ struct soc_mixer_control *mc = (struct soc_mixer_control *) kctl->private_value;
+ int vola, volb;
+ int ret;
+
+ ret = snd_soc_component_read(component, CS42L84_DAC_CHA_VOL_LSB);
+ if (ret < 0)
+ return ret;
+ vola = ret;
+
+ ret = snd_soc_component_read(component, CS42L84_DAC_CHA_VOL_MSB);
+ if (ret < 0)
+ return ret;
+ vola |= (ret & 1) << 8;
+
+ ret = snd_soc_component_read(component, CS42L84_DAC_CHB_VOL_LSB);
+ if (ret < 0)
+ return ret;
+ volb = ret;
+
+ ret = snd_soc_component_read(component, CS42L84_DAC_CHB_VOL_MSB);
+ if (ret < 0)
+ return ret;
+ volb |= (ret & 1) << 8;
+
+ if (vola & BIT(8))
+ vola |= ~((int)(BIT(8) - 1));
+ if (volb & BIT(8))
+ volb |= ~((int)(BIT(8) - 1));
+
+ val->value.integer.value[0] = vola - mc->min;
+ val->value.integer.value[1] = volb - mc->min;
+
+ return 0;
+}
+
+/* TODO */
+static const DECLARE_TLV_DB_SCALE(cs42l84_dac_tlv, -12800, 50, true);
+
+static const struct snd_kcontrol_new cs42l84_snd_controls[] = {
+ SOC_DOUBLE_R_S_EXT_TLV("DAC Playback Volume", CS42L84_DAC_CHA_VOL_LSB,
+ CS42L84_DAC_CHB_VOL_LSB, 0, -256, 24, 8, 0,
+ cs42l84_get_dac_vol, cs42l84_put_dac_vol, cs42l84_dac_tlv),
+ SOC_SINGLE("ADC Preamp Gain", CS42L84_ADC_CTL1,
+ CS42L84_ADC_CTL1_PREAMP_GAIN_SHIFT, 2, 0),
+ SOC_SINGLE("ADC PGA Gain", CS42L84_ADC_CTL1,
+ CS42L84_ADC_CTL1_PGA_GAIN_SHIFT, 31, 0),
+ SOC_SINGLE("ADC WNF Switch", CS42L84_ADC_CTL4,
+ CS42L84_ADC_CTL4_WNF_EN_SHIFT, 1, 0),
+ SOC_SINGLE("WNF Corner Frequency", CS42L84_ADC_CTL4,
+ CS42L84_ADC_CTL4_WNF_CF_SHIFT, 3, 0),
+ SOC_SINGLE("ADC HPF Switch", CS42L84_ADC_CTL4,
+ CS42L84_ADC_CTL4_HPF_EN_SHIFT, 1, 0),
+ SOC_SINGLE("HPF Corner Frequency", CS42L84_ADC_CTL4,
+ CS42L84_ADC_CTL4_HPF_CF_SHIFT, 3, 0),
+};
+
+static const char* const cs42l84_mux_text[] = {
+ "Blank", "ADC", "ASP RX CH1", "ASP RX CH2",
+};
+
+static const unsigned int cs42l84_mux_values[] = {
+ 0b0000, 0b0111, 0b1101, 0b1110,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(cs42l84_daca_mux_enum,
+ CS42L84_BUS_DAC_SRC, CS42L84_BUS_DAC_SRC_DACA_SHIFT,
+ 0b1111, cs42l84_mux_text, cs42l84_mux_values);
+
+static SOC_VALUE_ENUM_SINGLE_DECL(cs42l84_dacb_mux_enum,
+ CS42L84_BUS_DAC_SRC, CS42L84_BUS_DAC_SRC_DACB_SHIFT,
+ 0b1111, cs42l84_mux_text, cs42l84_mux_values);
+
+static SOC_VALUE_ENUM_SINGLE_DECL(cs42l84_sdout1_mux_enum,
+ CS42L84_BUS_ASP_TX_SRC, CS42L84_BUS_ASP_TX_SRC_CH1_SHIFT,
+ 0b1111, cs42l84_mux_text, cs42l84_mux_values);
+
+static const struct snd_kcontrol_new cs42l84_daca_mux_ctrl =
+ SOC_DAPM_ENUM("DACA Select", cs42l84_daca_mux_enum);
+
+static const struct snd_kcontrol_new cs42l84_dacb_mux_ctrl =
+ SOC_DAPM_ENUM("DACB Select", cs42l84_dacb_mux_enum);
+
+static const struct snd_kcontrol_new cs42l84_sdout1_mux_ctrl =
+ SOC_DAPM_ENUM("SDOUT1 Select", cs42l84_sdout1_mux_enum);
+
+static const struct snd_soc_dapm_widget cs42l84_dapm_widgets[] = {
+ /* Playback Path */
+ SND_SOC_DAPM_OUTPUT("HP"),
+ SND_SOC_DAPM_DAC("DAC", NULL, CS42L84_MSM_BLOCK_EN2, CS42L84_MSM_BLOCK_EN2_DAC_SHIFT, 0),
+ SND_SOC_DAPM_MUX("DACA Select", SND_SOC_NOPM, 0, 0, &cs42l84_daca_mux_ctrl),
+ SND_SOC_DAPM_MUX("DACB Select", SND_SOC_NOPM, 0, 0, &cs42l84_dacb_mux_ctrl),
+ SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, CS42L84_ASP_RX_EN, CS42L84_ASP_RX_EN_CH1_SHIFT, 0),
+ SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, CS42L84_ASP_RX_EN, CS42L84_ASP_RX_EN_CH2_SHIFT, 0),
+
+ /* Capture Path */
+ SND_SOC_DAPM_INPUT("HS"),
+ SND_SOC_DAPM_ADC("ADC", NULL, CS42L84_MSM_BLOCK_EN2, CS42L84_MSM_BLOCK_EN2_ADC_SHIFT, 0),
+ SND_SOC_DAPM_MUX("SDOUT1 Select", SND_SOC_NOPM, 0, 0, &cs42l84_sdout1_mux_ctrl),
+ SND_SOC_DAPM_AIF_OUT("SDOUT1", NULL, 0, CS42L84_ASP_TX_EN, CS42L84_ASP_TX_EN_CH1_SHIFT, 0),
+
+ /* Playback/Capture Requirements */
+ SND_SOC_DAPM_SUPPLY("BUS", CS42L84_MSM_BLOCK_EN2, CS42L84_MSM_BLOCK_EN2_BUS_SHIFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ASP", CS42L84_MSM_BLOCK_EN2, CS42L84_MSM_BLOCK_EN2_ASP_SHIFT, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("BCLK", CS42L84_ASP_CTL, CS42L84_ASP_CTL_BCLK_EN_SHIFT, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route cs42l84_audio_map[] = {
+ /* Playback Path */
+ {"HP", NULL, "DAC"},
+ {"DAC", NULL, "DACA Select"},
+ {"DAC", NULL, "DACB Select"},
+ {"DACA Select", "ASP RX CH1", "SDIN1"},
+ {"DACA Select", "ASP RX CH2", "SDIN2"},
+ {"DACB Select", "ASP RX CH1", "SDIN1"},
+ {"DACB Select", "ASP RX CH2", "SDIN2"},
+ {"SDIN1", NULL, "Playback"},
+ {"SDIN2", NULL, "Playback"},
+
+ {"ADC", NULL, "HS"},
+ {"SDOUT1 Select", "ADC", "ADC"},
+ {"SDOUT1", NULL, "SDOUT1 Select"},
+ {"Capture", NULL, "SDOUT1"},
+
+ /* Playback Requirements */
+ {"DAC", NULL, "BUS"},
+ {"SDIN1", NULL, "ASP"},
+ {"SDIN2", NULL, "ASP"},
+ {"SDIN1", NULL, "BCLK"},
+ {"SDIN2", NULL, "BCLK"},
+
+ /* Capture Requirements */
+ {"SDOUT1", NULL, "BUS"},
+ {"SDOUT1", NULL, "ASP"},
+ {"SDOUT1", NULL, "BCLK"},
+};
+
+static int cs42l84_set_jack(struct snd_soc_component *component, struct snd_soc_jack *jk, void *d)
+{
+ struct cs42l84_private *cs42l84 = snd_soc_component_get_drvdata(component);
+
+ /* Prevent race with interrupt handler */
+ mutex_lock(&cs42l84->irq_lock);
+ cs42l84->jack = jk;
+ snd_soc_jack_report(jk, cs42l84->hs_type, SND_JACK_HEADSET);
+ mutex_unlock(&cs42l84->irq_lock);
+
+ return 0;
+}
+
+static int cs42l84_component_probe(struct snd_soc_component *component)
+{
+ snd_soc_component_update_bits(component, CS42L84_ASP_CTL,
+ CS42L84_ASP_CTL_TDM_MODE, 0);
+ snd_soc_component_update_bits(component, CS42L84_HP_VOL_CTL,
+ CS42L84_HP_VOL_CTL_SOFT | CS42L84_HP_VOL_CTL_ZERO_CROSS,
+ CS42L84_HP_VOL_CTL_ZERO_CROSS);
+
+ /* TDM settings */
+ snd_soc_component_update_bits(component, CS42L84_ASP_RX_CH1_CTL1,
+ CS42L84_ASP_RX_CHx_CTL1_EDGE |
+ CS42L84_ASP_RX_CHx_CTL1_SLOT_START_LSB, 0);
+ snd_soc_component_update_bits(component, CS42L84_ASP_RX_CH1_CTL2,
+ CS42L84_ASP_RX_CHx_CTL2_SLOT_START_MSB, 0);
+ snd_soc_component_update_bits(component, CS42L84_ASP_RX_CH2_CTL1,
+ CS42L84_ASP_RX_CHx_CTL1_EDGE |
+ CS42L84_ASP_RX_CHx_CTL1_SLOT_START_LSB,
+ CS42L84_ASP_RX_CHx_CTL1_EDGE);
+ snd_soc_component_update_bits(component, CS42L84_ASP_RX_CH2_CTL2,
+ CS42L84_ASP_RX_CHx_CTL2_SLOT_START_MSB, 0);
+ snd_soc_component_update_bits(component, CS42L84_ASP_TX_CH1_CTL1,
+ CS42L84_ASP_RX_CHx_CTL1_EDGE | \
+ CS42L84_ASP_RX_CHx_CTL1_SLOT_START_LSB, 0);
+ snd_soc_component_update_bits(component, CS42L84_ASP_TX_CH1_CTL2,
+ CS42L84_ASP_RX_CHx_CTL2_SLOT_START_MSB, 0);
+ snd_soc_component_update_bits(component, CS42L84_ASP_TX_CH2_CTL1,
+ CS42L84_ASP_RX_CHx_CTL1_EDGE | \
+ CS42L84_ASP_RX_CHx_CTL1_SLOT_START_LSB,
+ CS42L84_ASP_RX_CHx_CTL1_EDGE);
+ snd_soc_component_update_bits(component, CS42L84_ASP_TX_CH2_CTL2,
+ CS42L84_ASP_RX_CHx_CTL2_SLOT_START_MSB, 0);
+ /* Routing defaults */
+ snd_soc_component_write(component, CS42L84_BUS_DAC_SRC,
+ 0b1101 << CS42L84_BUS_DAC_SRC_DACA_SHIFT |
+ 0b1110 << CS42L84_BUS_DAC_SRC_DACB_SHIFT);
+ snd_soc_component_write(component, CS42L84_BUS_ASP_TX_SRC,
+ 0b0111 << CS42L84_BUS_ASP_TX_SRC_CH1_SHIFT);
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver soc_component_dev_cs42l84 = {
+ .set_jack = cs42l84_set_jack,
+ .probe = cs42l84_component_probe,
+ .controls = cs42l84_snd_controls,
+ .num_controls = ARRAY_SIZE(cs42l84_snd_controls),
+ .dapm_widgets = cs42l84_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs42l84_dapm_widgets),
+ .dapm_routes = cs42l84_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cs42l84_audio_map),
+ .endianness = 1,
+};
+
+struct cs42l84_pll_params {
+ u32 bclk;
+ u8 mclk_src_sel;
+ u8 bclk_prediv;
+ u8 pll_div_int;
+ u32 pll_div_frac;
+ u8 pll_mode;
+ u8 pll_divout;
+ u32 mclk_int;
+};
+
+/*
+ * Common PLL Settings for given BCLK
+ */
+static const struct cs42l84_pll_params pll_ratio_table[] = {
+ { 3072000, 1, 0, 0x40, 0x000000, 0x03, 0x10, 12288000},
+ { 6144000, 1, 1, 0x40, 0x000000, 0x03, 0x10, 12288000},
+ { 12288000, 0, 0, 0, 0, 0, 0, 12288000},
+ { 24576000, 1, 3, 0x40, 0x000000, 0x03, 0x10, 12288000},
+};
+
+static int cs42l84_pll_config(struct snd_soc_component *component)
+{
+ struct cs42l84_private *cs42l84 = snd_soc_component_get_drvdata(component);
+ int i;
+ u32 clk;
+ u32 fsync;
+
+ clk = cs42l84->bclk;
+
+ /* Don't reconfigure if there is an audio stream running */
+ if (cs42l84->stream_use) {
+ if (pll_ratio_table[cs42l84->pll_config].bclk == clk)
+ return 0;
+ else
+ return -EBUSY;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
+ if (pll_ratio_table[i].bclk == clk) {
+ cs42l84->pll_config = i;
+ break;
+ }
+ }
+
+ if (i == ARRAY_SIZE(pll_ratio_table))
+ return -EINVAL;
+
+ /* Set up the LRCLK */
+ fsync = clk / cs42l84->srate;
+ if (((fsync * cs42l84->srate) != clk)
+ || ((fsync % 2) != 0)) {
+ dev_err(component->dev,
+ "Unsupported bclk %d/sample rate %d\n",
+ clk, cs42l84->srate);
+ return -EINVAL;
+ }
+
+ /* Set the LRCLK period */
+ snd_soc_component_update_bits(component, CS42L84_ASP_FSYNC_CTL2,
+ CS42L84_ASP_FSYNC_CTL2_BCLK_PERIOD_LO,
+ FIELD_PREP(CS42L84_ASP_FSYNC_CTL2_BCLK_PERIOD_LO, fsync & 0x7f));
+ snd_soc_component_update_bits(component, CS42L84_ASP_FSYNC_CTL3,
+ CS42L84_ASP_FSYNC_CTL3_BCLK_PERIOD_HI,
+ FIELD_PREP(CS42L84_ASP_FSYNC_CTL3_BCLK_PERIOD_HI, fsync >> 7));
+
+ /* Save what the MCLK will be */
+ switch (pll_ratio_table[i].mclk_int) {
+ case 12000000:
+ cs42l84->pll_mclk_f = CS42L84_CCM_CTL1_MCLK_F_12MHZ;
+ break;
+ case 12288000:
+ cs42l84->pll_mclk_f = CS42L84_CCM_CTL1_MCLK_F_12_288KHZ;
+ break;
+ case 24000000:
+ cs42l84->pll_mclk_f = CS42L84_CCM_CTL1_MCLK_F_24MHZ;
+ break;
+ case 24576000:
+ cs42l84->pll_mclk_f = CS42L84_CCM_CTL1_MCLK_F_24_576KHZ;
+ break;
+ }
+
+ if (pll_ratio_table[i].mclk_src_sel) {
+ /* Configure PLL */
+ snd_soc_component_update_bits(component,
+ CS42L84_CCM_CTL3, CS42L84_CCM_CTL3_REFCLK_DIV,
+ FIELD_PREP(CS42L84_CCM_CTL3_REFCLK_DIV, pll_ratio_table[i].bclk_prediv));
+ snd_soc_component_write(component,
+ CS42L84_PLL_DIV_INT,
+ pll_ratio_table[i].pll_div_int);
+ snd_soc_component_write(component,
+ CS42L84_PLL_DIV_FRAC0,
+ pll_ratio_table[i].pll_div_frac);
+ snd_soc_component_write(component,
+ CS42L84_PLL_DIV_FRAC1,
+ pll_ratio_table[i].pll_div_frac >> 8);
+ snd_soc_component_write(component,
+ CS42L84_PLL_DIV_FRAC2,
+ pll_ratio_table[i].pll_div_frac >> 16);
+ snd_soc_component_update_bits(component,
+ CS42L84_PLL_CTL1, CS42L84_PLL_CTL1_MODE,
+ FIELD_PREP(CS42L84_PLL_CTL1_MODE, pll_ratio_table[i].pll_mode));
+ snd_soc_component_write(component,
+ CS42L84_PLL_DIVOUT,
+ pll_ratio_table[i].pll_divout);
+
+ snd_soc_component_update_bits(component,
+ CS42L84_PLL_CTL1, CS42L84_PLL_CTL1_EN,
+ CS42L84_PLL_CTL1_EN);
+ }
+
+ return 0;
+}
+
+static int cs42l84_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
+ case SND_SOC_DAIFMT_BC_FC:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Bitclock/frame inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int cs42l84_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct cs42l84_private *cs42l84 = snd_soc_component_get_drvdata(component);
+ int ret;
+ u32 ccm_samp_rate;
+
+ cs42l84->srate = params_rate(params);
+
+ ret = cs42l84_pll_config(component);
+ if (ret)
+ return ret;
+
+ switch (params_rate(params)) {
+ case 44100:
+ ccm_samp_rate = CS42L84_CCM_SAMP_RATE_RATE_44K1HZ;
+ break;
+ case 48000:
+ ccm_samp_rate = CS42L84_CCM_SAMP_RATE_RATE_48KHZ;
+ break;
+ case 88200:
+ ccm_samp_rate = CS42L84_CCM_SAMP_RATE_RATE_88K2HZ;
+ break;
+ case 96000:
+ ccm_samp_rate = CS42L84_CCM_SAMP_RATE_RATE_96KHZ;
+ break;
+ case 176400:
+ ccm_samp_rate = CS42L84_CCM_SAMP_RATE_RATE_176K4HZ;
+ break;
+ case 192000:
+ ccm_samp_rate = CS42L84_CCM_SAMP_RATE_RATE_192KHZ;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_component_write(component, CS42L84_CCM_SAMP_RATE, ccm_samp_rate);
+
+ switch (substream->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ snd_soc_component_write(component, CS42L84_ASP_RX_CH1_WIDTH,
+ params_width(params) - 1);
+ snd_soc_component_write(component, CS42L84_ASP_RX_CH2_WIDTH,
+ params_width(params) - 1);
+ break;
+
+ case SNDRV_PCM_STREAM_CAPTURE:
+ snd_soc_component_write(component, CS42L84_ASP_TX_CH1_WIDTH,
+ params_width(params) - 1);
+ snd_soc_component_write(component, CS42L84_ASP_TX_CH2_WIDTH,
+ params_width(params) - 1);
+ break;
+ }
+
+ return 0;
+}
+
+static int cs42l84_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_component *component = dai->component;
+ struct cs42l84_private *cs42l84 = snd_soc_component_get_drvdata(component);
+ int i;
+
+ if (freq == 0) {
+ cs42l84->bclk = 0;
+ return 0;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
+ if (pll_ratio_table[i].bclk == freq) {
+ cs42l84->bclk = freq;
+ return 0;
+ }
+ }
+
+ dev_err(component->dev, "BCLK %u not supported\n", freq);
+
+ return -EINVAL;
+}
+
+static int cs42l84_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
+{
+ struct snd_soc_component *component = dai->component;
+ struct cs42l84_private *cs42l84 = snd_soc_component_get_drvdata(component);
+ unsigned int regval;
+ int ret;
+
+ if (mute) {
+ /* Mute the headphone */
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_component_update_bits(component, CS42L84_DAC_CTL1,
+ CS42L84_DAC_CTL1_UNMUTE, 0);
+ cs42l84->stream_use &= ~(1 << stream);
+ if (!cs42l84->stream_use) {
+ /* Must disconnect PLL before stopping it */
+ snd_soc_component_write(component, CS42L84_CCM_CTL1,
+ CS42L84_CCM_CTL1_RCO);
+
+ usleep_range(150, 300);
+
+ snd_soc_component_update_bits(component, CS42L84_PLL_CTL1,
+ CS42L84_PLL_CTL1_EN, 0);
+
+ snd_soc_component_update_bits(component, CS42L84_CCM_CTL4,
+ CS42L84_CCM_CTL4_REFCLK_EN, 0);
+ }
+ } else {
+ if (!cs42l84->stream_use) {
+ /* SCLK must be running before codec unmute.
+ *
+ * Note carried over from CS42L42:
+ *
+ * PLL must not be started with ADC and HP both off
+ * otherwise the FILT+ supply will not charge properly.
+ * DAPM widgets power-up before stream unmute so at least
+ * one of the "DAC" or "ADC" widgets will already have
+ * powered-up.
+ */
+
+ snd_soc_component_update_bits(component, CS42L84_CCM_CTL4,
+ CS42L84_CCM_CTL4_REFCLK_EN,
+ CS42L84_CCM_CTL4_REFCLK_EN);
+
+ if (pll_ratio_table[cs42l84->pll_config].mclk_src_sel) {
+ snd_soc_component_update_bits(component, CS42L84_PLL_CTL1,
+ CS42L84_PLL_CTL1_EN,
+ CS42L84_PLL_CTL1_EN);
+ /* TODO: should we be doing something with divout here? */
+
+ ret = regmap_read_poll_timeout(cs42l84->regmap,
+ CS42L84_PLL_LOCK_STATUS,
+ regval,
+ (regval & CS42L84_PLL_LOCK_STATUS_LOCKED),
+ CS42L84_PLL_LOCK_POLL_US,
+ CS42L84_PLL_LOCK_TIMEOUT_US);
+ if (ret < 0)
+ dev_warn(component->dev, "PLL failed to lock: %d\n", ret);
+
+ /* PLL must be running to drive glitchless switch logic */
+ snd_soc_component_update_bits(component,
+ CS42L84_CCM_CTL1,
+ CS42L84_CCM_CTL1_MCLK_SRC | CS42L84_CCM_CTL1_MCLK_FREQ,
+ FIELD_PREP(CS42L84_CCM_CTL1_MCLK_SRC, CS42L84_CCM_CTL1_MCLK_SRC_PLL)
+ | FIELD_PREP(CS42L84_CCM_CTL1_MCLK_FREQ, cs42l84->pll_mclk_f));
+ usleep_range(CS42L84_CLOCK_SWITCH_DELAY_US, CS42L84_CLOCK_SWITCH_DELAY_US*2);
+ } else {
+ snd_soc_component_update_bits(component,
+ CS42L84_CCM_CTL1,
+ CS42L84_CCM_CTL1_MCLK_SRC | CS42L84_CCM_CTL1_MCLK_FREQ,
+ FIELD_PREP(CS42L84_CCM_CTL1_MCLK_SRC, CS42L84_CCM_CTL1_MCLK_SRC_BCLK)
+ | FIELD_PREP(CS42L84_CCM_CTL1_MCLK_FREQ, cs42l84->pll_mclk_f));
+ usleep_range(CS42L84_CLOCK_SWITCH_DELAY_US, CS42L84_CLOCK_SWITCH_DELAY_US*2);
+ }
+ }
+ cs42l84->stream_use |= 1 << stream;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ /* Un-mute the headphone */
+ snd_soc_component_update_bits(component, CS42L84_DAC_CTL1,
+ CS42L84_DAC_CTL1_UNMUTE,
+ CS42L84_DAC_CTL1_UNMUTE);
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops cs42l84_ops = {
+ .hw_params = cs42l84_pcm_hw_params,
+ .set_fmt = cs42l84_set_dai_fmt,
+ .set_sysclk = cs42l84_set_sysclk,
+ .mute_stream = cs42l84_mute_stream,
+};
+
+#define CS42L84_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver cs42l84_dai = {
+ .name = "cs42l84",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000,
+ .formats = CS42L84_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000,
+ .formats = CS42L84_FORMATS,
+ },
+ .symmetric_rate = 1,
+ .symmetric_sample_bits = 1,
+ .ops = &cs42l84_ops,
+};
+
+struct cs42l84_irq_params {
+ u16 status_addr;
+ u16 mask_addr;
+ u8 mask;
+};
+
+static const struct cs42l84_irq_params irq_params_table[] = {
+ {CS42L84_TSRS_PLUG_INT_STATUS, CS42L84_TSRS_PLUG_INT_MASK,
+ CS42L84_TSRS_PLUG_VAL_MASK}
+};
+
+static void cs42l84_detect_hs(struct cs42l84_private *cs42l84)
+{
+ unsigned int reg;
+
+ /* Power up HSBIAS */
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_MISC_DET_CTL,
+ CS42L84_MISC_DET_CTL_HSBIAS_CTL | CS42L84_MISC_DET_CTL_DETECT_MODE,
+ FIELD_PREP(CS42L84_MISC_DET_CTL_HSBIAS_CTL, 3) | /* 2.7 V */
+ FIELD_PREP(CS42L84_MISC_DET_CTL_DETECT_MODE, 0));
+
+ /* Power up level detection circuitry */
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_MISC_DET_CTL,
+ CS42L84_MISC_DET_CTL_PDN_MIC_LVL_DET, 0);
+
+ /* TODO: Optimize */
+ msleep(100);
+
+ /* Connect HSBIAS in CTIA wiring */
+ /* TODO: Should likely be subject of detection */
+ regmap_write(cs42l84->regmap,
+ CS42L84_HS_SWITCH_CTL,
+ CS42L84_HS_SWITCH_CTL_REF_HS3 | \
+ CS42L84_HS_SWITCH_CTL_HSB_FILT_HS3 | \
+ CS42L84_HS_SWITCH_CTL_GNDHS_HS3 | \
+ CS42L84_HS_SWITCH_CTL_HSB_HS4);
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_HS_DET_CTL2,
+ CS42L84_HS_DET_CTL2_SET,
+ FIELD_PREP(CS42L84_HS_DET_CTL2_SET, 0));
+
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_MISC_DET_CTL,
+ CS42L84_MISC_DET_CTL_DETECT_MODE,
+ FIELD_PREP(CS42L84_MISC_DET_CTL_DETECT_MODE, 3));
+
+ /* TODO: Optimize */
+ msleep(100);
+
+ regmap_read(cs42l84->regmap, CS42L84_HS_DET_STATUS2, &reg);
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_MISC_DET_CTL,
+ CS42L84_MISC_DET_CTL_PDN_MIC_LVL_DET,
+ CS42L84_MISC_DET_CTL_PDN_MIC_LVL_DET);
+
+ switch (reg & 0b11) {
+ case 0b11: /* shorted */
+ case 0b00: /* open */
+ /* Power down HSBIAS */
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_MISC_DET_CTL,
+ CS42L84_MISC_DET_CTL_HSBIAS_CTL,
+ FIELD_PREP(CS42L84_MISC_DET_CTL_HSBIAS_CTL, 1)); /* 0.0 V */
+ break;
+ }
+
+ switch (reg & 0b11) {
+ case 0b10: /* load */
+ dev_dbg(cs42l84->dev, "Detected mic\n");
+ cs42l84->hs_type = SND_JACK_HEADSET;
+ snd_soc_jack_report(cs42l84->jack, SND_JACK_HEADSET,
+ SND_JACK_HEADSET);
+ break;
+
+ case 0b00: /* open */
+ dev_dbg(cs42l84->dev, "Detected open circuit on HS4\n");
+ fallthrough;
+ case 0b11: /* shorted */
+ default:
+ snd_soc_jack_report(cs42l84->jack, SND_JACK_HEADPHONE,
+ SND_JACK_HEADSET);
+ cs42l84->hs_type = SND_JACK_HEADPHONE;
+ dev_dbg(cs42l84->dev, "Detected bare headphone (no mic)\n");
+ }
+}
+
+static void cs42l84_revert_hs(struct cs42l84_private *cs42l84)
+{
+ /* Power down HSBIAS */
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_MISC_DET_CTL,
+ CS42L84_MISC_DET_CTL_HSBIAS_CTL | CS42L84_MISC_DET_CTL_DETECT_MODE,
+ FIELD_PREP(CS42L84_MISC_DET_CTL_HSBIAS_CTL, 1) | /* 0.0 V */
+ FIELD_PREP(CS42L84_MISC_DET_CTL_DETECT_MODE, 0));
+
+ /* Disconnect HSBIAS */
+ regmap_write(cs42l84->regmap,
+ CS42L84_HS_SWITCH_CTL,
+ CS42L84_HS_SWITCH_CTL_REF_HS3 | \
+ CS42L84_HS_SWITCH_CTL_REF_HS4 | \
+ CS42L84_HS_SWITCH_CTL_HSB_FILT_HS3 | \
+ CS42L84_HS_SWITCH_CTL_HSB_FILT_HS4 | \
+ CS42L84_HS_SWITCH_CTL_GNDHS_HS3 | \
+ CS42L84_HS_SWITCH_CTL_GNDHS_HS4);
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_HS_DET_CTL2,
+ CS42L84_HS_DET_CTL2_SET,
+ FIELD_PREP(CS42L84_HS_DET_CTL2_SET, 2));
+}
+
+static irqreturn_t cs42l84_irq_thread(int irq, void *data)
+{
+ struct cs42l84_private *cs42l84 = (struct cs42l84_private *)data;
+ unsigned int stickies[1];
+ unsigned int masks[1];
+ unsigned int reg;
+ u8 current_plug_status;
+ int i;
+
+ mutex_lock(&cs42l84->irq_lock);
+ /* Read sticky registers to clear interurpt */
+ for (i = 0; i < ARRAY_SIZE(stickies); i++) {
+ regmap_read(cs42l84->regmap, irq_params_table[i].status_addr,
+ &(stickies[i]));
+ regmap_read(cs42l84->regmap, irq_params_table[i].mask_addr,
+ &(masks[i]));
+ stickies[i] = stickies[i] & (~masks[i]) &
+ irq_params_table[i].mask;
+ }
+
+ if ((~masks[0]) & irq_params_table[0].mask) {
+ regmap_read(cs42l84->regmap, CS42L84_TSRS_PLUG_STATUS, &reg);
+ current_plug_status = (((char) reg) &
+ (CS42L84_TS_PLUG | CS42L84_TS_UNPLUG)) >>
+ CS42L84_TS_PLUG_SHIFT;
+
+ switch (current_plug_status) {
+ case CS42L84_PLUG:
+ if (cs42l84->plug_state != CS42L84_PLUG) {
+ cs42l84->plug_state = CS42L84_PLUG;
+ dev_dbg(cs42l84->dev, "Plug event\n");
+
+ cs42l84_detect_hs(cs42l84);
+
+ /*
+ * Check the tip sense status again, and possibly invalidate
+ * the detection result
+ *
+ * Thanks to debounce, this should reliably indicate if the tip
+ * was disconnected at any point during the detection procedure.
+ */
+ regmap_read(cs42l84->regmap, CS42L84_TSRS_PLUG_STATUS, &reg);
+ current_plug_status = (((char) reg) &
+ (CS42L84_TS_PLUG | CS42L84_TS_UNPLUG)) >>
+ CS42L84_TS_PLUG_SHIFT;
+ if (current_plug_status != CS42L84_PLUG) {
+ dev_dbg(cs42l84->dev, "Wobbly connection, detection invalidated\n");
+ cs42l84->plug_state = CS42L84_UNPLUG;
+ cs42l84_revert_hs(cs42l84);
+ }
+ }
+ break;
+
+ case CS42L84_UNPLUG:
+ if (cs42l84->plug_state != CS42L84_UNPLUG) {
+ cs42l84->plug_state = CS42L84_UNPLUG;
+ dev_dbg(cs42l84->dev, "Unplug event\n");
+
+ cs42l84_revert_hs(cs42l84);
+ cs42l84->hs_type = 0;
+ snd_soc_jack_report(cs42l84->jack, 0,
+ SND_JACK_HEADSET);
+ }
+ break;
+
+ default:
+ if (cs42l84->plug_state != CS42L84_TRANS)
+ cs42l84->plug_state = CS42L84_TRANS;
+ }
+ }
+ mutex_unlock(&cs42l84->irq_lock);
+
+ return IRQ_HANDLED;
+}
+
+static void cs42l84_set_interrupt_masks(struct cs42l84_private *cs42l84)
+{
+ regmap_update_bits(cs42l84->regmap, CS42L84_TSRS_PLUG_INT_MASK,
+ CS42L84_RS_PLUG | CS42L84_RS_UNPLUG |
+ CS42L84_TS_PLUG | CS42L84_TS_UNPLUG,
+ CS42L84_RS_PLUG | CS42L84_RS_UNPLUG);
+}
+
+static void cs42l84_setup_plug_detect(struct cs42l84_private *cs42l84)
+{
+ unsigned int reg;
+
+ /* Set up plug detection */
+ regmap_update_bits(cs42l84->regmap, CS42L84_MIC_DET_CTL4,
+ CS42L84_MIC_DET_CTL4_LATCH_TO_VP,
+ CS42L84_MIC_DET_CTL4_LATCH_TO_VP);
+ regmap_update_bits(cs42l84->regmap, CS42L84_TIP_SENSE_CTL2,
+ CS42L84_TIP_SENSE_CTL2_MODE,
+ FIELD_PREP(CS42L84_TIP_SENSE_CTL2_MODE, CS42L84_TIP_SENSE_CTL2_MODE_SHORT_DET));
+ regmap_update_bits(cs42l84->regmap, CS42L84_RING_SENSE_CTL,
+ CS42L84_RING_SENSE_CTL_INV | CS42L84_RING_SENSE_CTL_UNK1 |
+ CS42L84_RING_SENSE_CTL_RISETIME | CS42L84_RING_SENSE_CTL_FALLTIME,
+ CS42L84_RING_SENSE_CTL_INV | CS42L84_RING_SENSE_CTL_UNK1 |
+ FIELD_PREP(CS42L84_RING_SENSE_CTL_RISETIME, CS42L84_DEBOUNCE_TIME_125MS) |
+ FIELD_PREP(CS42L84_RING_SENSE_CTL_FALLTIME, CS42L84_DEBOUNCE_TIME_125MS));
+ regmap_update_bits(cs42l84->regmap, CS42L84_TIP_SENSE_CTL,
+ CS42L84_TIP_SENSE_CTL_INV |
+ CS42L84_TIP_SENSE_CTL_RISETIME | CS42L84_TIP_SENSE_CTL_FALLTIME,
+ CS42L84_TIP_SENSE_CTL_INV |
+ FIELD_PREP(CS42L84_TIP_SENSE_CTL_RISETIME, CS42L84_DEBOUNCE_TIME_500MS) |
+ FIELD_PREP(CS42L84_TIP_SENSE_CTL_FALLTIME, CS42L84_DEBOUNCE_TIME_125MS));
+ regmap_update_bits(cs42l84->regmap, CS42L84_MSM_BLOCK_EN3,
+ CS42L84_MSM_BLOCK_EN3_TR_SENSE,
+ CS42L84_MSM_BLOCK_EN3_TR_SENSE);
+
+ /* Save the initial status of the tip sense */
+ regmap_read(cs42l84->regmap, CS42L84_TSRS_PLUG_STATUS, &reg);
+ cs42l84->plug_state = (((char) reg) &
+ (CS42L84_TS_PLUG | CS42L84_TS_UNPLUG)) >>
+ CS42L84_TS_PLUG_SHIFT;
+
+ /* Set mic-detection threshold */
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_MIC_DET_CTL1, CS42L84_MIC_DET_CTL1_HS_DET_LEVEL,
+ FIELD_PREP(CS42L84_MIC_DET_CTL1_HS_DET_LEVEL, 0x2c)); /* ~1.9 V */
+
+ /* Disconnect HSBIAS (initially) */
+ regmap_write(cs42l84->regmap,
+ CS42L84_HS_SWITCH_CTL,
+ CS42L84_HS_SWITCH_CTL_REF_HS3 | \
+ CS42L84_HS_SWITCH_CTL_REF_HS4 | \
+ CS42L84_HS_SWITCH_CTL_HSB_FILT_HS3 | \
+ CS42L84_HS_SWITCH_CTL_HSB_FILT_HS4 | \
+ CS42L84_HS_SWITCH_CTL_GNDHS_HS3 | \
+ CS42L84_HS_SWITCH_CTL_GNDHS_HS4);
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_HS_DET_CTL2,
+ CS42L84_HS_DET_CTL2_SET | CS42L84_HS_DET_CTL2_CTL,
+ FIELD_PREP(CS42L84_HS_DET_CTL2_SET, 2) |
+ FIELD_PREP(CS42L84_HS_DET_CTL2_CTL, 0));
+ regmap_update_bits(cs42l84->regmap,
+ CS42L84_HS_CLAMP_DISABLE, 1, 1);
+
+}
+
+static int cs42l84_i2c_probe(struct i2c_client *i2c_client,
+ const struct i2c_device_id *id)
+{
+ struct cs42l84_private *cs42l84;
+ int ret, devid;
+ unsigned int reg;
+
+ cs42l84 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l84_private),
+ GFP_KERNEL);
+ if (!cs42l84)
+ return -ENOMEM;
+
+ cs42l84->dev = &i2c_client->dev;
+ i2c_set_clientdata(i2c_client, cs42l84);
+ mutex_init(&cs42l84->irq_lock);
+
+ cs42l84->regmap = devm_regmap_init_i2c(i2c_client, &cs42l84_regmap);
+ if (IS_ERR(cs42l84->regmap)) {
+ ret = PTR_ERR(cs42l84->regmap);
+ dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+ return ret;
+ }
+
+ /* Reset the Device */
+ cs42l84->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev,
+ "reset", GPIOD_OUT_LOW);
+ if (IS_ERR(cs42l84->reset_gpio)) {
+ ret = PTR_ERR(cs42l84->reset_gpio);
+ goto err_disable_noreset;
+ }
+
+ if (cs42l84->reset_gpio) {
+ dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
+ gpiod_set_value_cansleep(cs42l84->reset_gpio, 1);
+ }
+ usleep_range(CS42L84_BOOT_TIME_US, CS42L84_BOOT_TIME_US * 2);
+
+ /* Request IRQ if one was specified */
+ if (i2c_client->irq) {
+ ret = request_threaded_irq(i2c_client->irq,
+ NULL, cs42l84_irq_thread,
+ IRQF_ONESHOT,
+ "cs42l84", cs42l84);
+ if (ret == -EPROBE_DEFER) {
+ goto err_disable_noirq;
+ } else if (ret != 0) {
+ dev_err(&i2c_client->dev,
+ "Failed to request IRQ: %d\n", ret);
+ goto err_disable_noirq;
+ }
+ }
+
+ /* initialize codec */
+ devid = cirrus_read_device_id(cs42l84->regmap, CS42L84_DEVID);
+ if (devid < 0) {
+ ret = devid;
+ dev_err(&i2c_client->dev, "Failed to read device ID: %d\n", ret);
+ goto err_disable;
+ }
+
+ if (devid != CS42L84_CHIP_ID) {
+ dev_err(&i2c_client->dev,
+ "CS42L84 Device ID (%X). Expected %X\n",
+ devid, CS42L84_CHIP_ID);
+ ret = -EINVAL;
+ goto err_disable;
+ }
+
+ ret = regmap_read(cs42l84->regmap, CS42L84_REVID, &reg);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "Get Revision ID failed\n");
+ goto err_shutdown;
+ }
+
+ dev_info(&i2c_client->dev,
+ "Cirrus Logic CS42L84, Revision: %02X\n", reg & 0xFF);
+
+ /* Setup plug detection */
+ cs42l84_setup_plug_detect(cs42l84);
+
+ /* Mask/Unmask Interrupts */
+ cs42l84_set_interrupt_masks(cs42l84);
+
+ /* Register codec for machine driver */
+ ret = devm_snd_soc_register_component(&i2c_client->dev,
+ &soc_component_dev_cs42l84, &cs42l84_dai, 1);
+ if (ret < 0)
+ goto err_shutdown;
+
+ return 0;
+
+err_shutdown:
+ /* Nothing to do */
+
+err_disable:
+ if (i2c_client->irq)
+ free_irq(i2c_client->irq, cs42l84);
+
+err_disable_noirq:
+ gpiod_set_value_cansleep(cs42l84->reset_gpio, 0);
+err_disable_noreset:
+ return ret;
+}
+
+static void cs42l84_i2c_remove(struct i2c_client *i2c_client)
+{
+ struct cs42l84_private *cs42l84 = i2c_get_clientdata(i2c_client);
+
+ if (i2c_client->irq)
+ free_irq(i2c_client->irq, cs42l84);
+
+ gpiod_set_value_cansleep(cs42l84->reset_gpio, 0);
+}
+
+static const struct of_device_id cs42l84_of_match[] = {
+ { .compatible = "cirrus,cs42l84", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, cs42l84_of_match);
+
+static const struct i2c_device_id cs42l84_id[] = {
+ {"cs42l84", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, cs42l84_id);
+
+static struct i2c_driver cs42l84_i2c_driver = {
+ .driver = {
+ .name = "cs42l84",
+ .of_match_table = of_match_ptr(cs42l84_of_match),
+ },
+ .id_table = cs42l84_id,
+ .probe = cs42l84_i2c_probe,
+ .remove = cs42l84_i2c_remove,
+};
+
+module_i2c_driver(cs42l84_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS42L84 driver");
+MODULE_AUTHOR("Martin PoviĊĦer <povik+lin@cutebit.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l84.h b/sound/soc/codecs/cs42l84.h
new file mode 100644
index 000000000000..9aaf19051d39
--- /dev/null
+++ b/sound/soc/codecs/cs42l84.h
@@ -0,0 +1,217 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * Copyright (C) The Asahi Linux Contributors
+ *
+ * Based on sound/soc/codecs/cs42l42.h
+ *
+ * Copyright 2016 Cirrus Logic, Inc.
+ */
+
+
+#ifndef __CS42L84_H__
+#define __CS42L84_H__
+
+#include <linux/bits.h>
+
+#define CS42L84_CHIP_ID 0x42a84
+
+#define CS42L84_DEVID 0x0000
+#define CS42L84_REVID 0x73fe
+#define CS42L84_FRZ_CTL 0x0006
+#define CS42L84_FRZ_CTL_ENGAGE BIT(0)
+
+#define CS42L84_TSRS_PLUG_INT_STATUS 0x0400
+#define CS42L84_TSRS_PLUG_INT_MASK 0x0418
+#define CS42L84_RS_PLUG_SHIFT 0
+#define CS42L84_RS_PLUG BIT(0)
+#define CS42L84_RS_UNPLUG BIT(1)
+#define CS42L84_TS_PLUG_SHIFT 2
+#define CS42L84_TS_PLUG BIT(2)
+#define CS42L84_TS_UNPLUG BIT(3)
+#define CS42L84_TSRS_PLUG_VAL_MASK GENMASK(3, 0)
+#define CS42L84_PLL_LOCK_STATUS 0x040e // probably bit 0x10
+#define CS42L84_PLL_LOCK_STATUS_LOCKED BIT(4)
+
+#define CS42L84_PLUG 3
+#define CS42L84_UNPLUG 0
+#define CS42L84_TRANS 1
+
+#if 0
+ l84.regs.RING_SENSE_CTRL.set(INV=1, UNK1=1,
+ RISETIME=E_DEBOUNCE_TIME.T_125MS, FALLTIME=E_DEBOUNCE_TIME.T_125MS)
+ l84.regs.TIP_SENSE_CTRL.set(INV=1,
+ RISETIME=E_DEBOUNCE_TIME.T_500MS, FALLTIME=E_DEBOUNCE_TIME.T_125MS)
+ l84.regs.MSM_BLOCK_EN3.set(TR_SENSE_EN=1)
+#endif
+
+#define CS42L84_CCM_CTL1 0x0600
+#define CS42L84_CCM_CTL1_MCLK_SRC GENMASK(1, 0)
+#define CS42L84_CCM_CTL1_MCLK_SRC_RCO 0
+#define CS42L84_CCM_CTL1_MCLK_SRC_MCLK 1
+#define CS42L84_CCM_CTL1_MCLK_SRC_BCLK 2
+#define CS42L84_CCM_CTL1_MCLK_SRC_PLL 3
+#define CS42L84_CCM_CTL1_MCLK_FREQ GENMASK(3, 2)
+#define CS42L84_CCM_CTL1_MCLK_F_12MHZ 0b00
+#define CS42L84_CCM_CTL1_MCLK_F_24MHZ 0b01
+#define CS42L84_CCM_CTL1_MCLK_F_12_288KHZ 0b10
+#define CS42L84_CCM_CTL1_MCLK_F_24_576KHZ 0b11
+#define CS42L84_CCM_CTL1_RCO \
+ (FIELD_PREP(CS42L84_CCM_CTL1_MCLK_SRC, CS42L84_CCM_CTL1_MCLK_SRC_RCO) \
+ | FIELD_PREP(CS42L84_CCM_CTL1_MCLK_FREQ, CS42L84_CCM_CTL1_MCLK_F_12MHZ))
+
+#define CS42L84_CCM_SAMP_RATE 0x0601
+#define CS42L84_CCM_SAMP_RATE_RATE_48KHZ 4
+#define CS42L84_CCM_SAMP_RATE_RATE_96KHZ 5
+#define CS42L84_CCM_SAMP_RATE_RATE_192KHZ 6
+#define CS42L84_CCM_SAMP_RATE_RATE_44K1HZ 12
+#define CS42L84_CCM_SAMP_RATE_RATE_88K2HZ 13
+#define CS42L84_CCM_SAMP_RATE_RATE_176K4HZ 14
+#define CS42L84_CCM_CTL3 0x0602
+#define CS42L84_CCM_CTL3_REFCLK_DIV GENMASK(2, 1)
+#define CS42L84_CCM_CTL4 0x0603
+#define CS42L84_CCM_CTL4_REFCLK_EN BIT(0)
+
+#define CS42L84_CCM_ASP_CLK_CTRL 0x0608
+
+#define CS42L84_PLL_CTL1 0x0800
+#define CS42L84_PLL_CTL1_EN BIT(0)
+#define CS42L84_PLL_CTL1_MODE GENMASK(2, 1)
+#define CS42L84_PLL_DIV_FRAC0 0x0804
+#define CS42L84_PLL_DIV_FRAC1 0x0805
+#define CS42L84_PLL_DIV_FRAC2 0x0806
+#define CS42L84_PLL_DIV_INT 0x0807
+#define CS42L84_PLL_DIVOUT 0x0808
+
+#define CS42L84_RING_SENSE_CTL 0x1282
+#define CS42L84_RING_SENSE_CTL_INV BIT(7)
+#define CS42L84_RING_SENSE_CTL_UNK1 BIT(6)
+#define CS42L84_RING_SENSE_CTL_FALLTIME GENMASK(5, 3)
+#define CS42L84_RING_SENSE_CTL_RISETIME GENMASK(2, 0)
+#define CS42L84_TIP_SENSE_CTL 0x1283
+#define CS42L84_TIP_SENSE_CTL_INV BIT(7)
+#define CS42L84_TIP_SENSE_CTL_FALLTIME GENMASK(5, 3)
+#define CS42L84_TIP_SENSE_CTL_RISETIME GENMASK(2, 0)
+
+#define CS42L84_TSRS_PLUG_STATUS 0x1288
+
+#define CS42L84_TIP_SENSE_CTL2 0x1473
+#define CS42L84_TIP_SENSE_CTL2_MODE GENMASK(7, 6)
+#define CS42L84_TIP_SENSE_CTL2_MODE_DISABLED 0b00
+#define CS42L84_TIP_SENSE_CTL2_MODE_DIG_INPUT 0b01
+#define CS42L84_TIP_SENSE_CTL2_MODE_SHORT_DET 0b11
+#define CS42L84_TIP_SENSE_CTL2_INV BIT(5)
+
+#define CS42L84_MISC_DET_CTL 0x1474
+#define CS42L84_MISC_DET_CTL_DETECT_MODE GENMASK(4, 3)
+#define CS42L84_MISC_DET_CTL_HSBIAS_CTL GENMASK(2, 1)
+#define CS42L84_MISC_DET_CTL_PDN_MIC_LVL_DET BIT(0)
+
+#define CS42L84_MIC_DET_CTL1 0x1475
+#define CS42L84_MIC_DET_CTL1_HS_DET_LEVEL GENMASK(5, 0)
+
+#define CS42L84_MIC_DET_CTL4 0x1477
+#define CS42L84_MIC_DET_CTL4_LATCH_TO_VP BIT(1)
+
+#define CS42L84_HS_DET_STATUS2 0x147d
+
+#define CS42L84_MSM_BLOCK_EN1 0x1800
+#define CS42L84_MSM_BLOCK_EN2 0x1801
+#define CS42L84_MSM_BLOCK_EN2_ASP_SHIFT 6
+#define CS42L84_MSM_BLOCK_EN2_BUS_SHIFT 5
+#define CS42L84_MSM_BLOCK_EN2_DAC_SHIFT 4
+#define CS42L84_MSM_BLOCK_EN2_ADC_SHIFT 3
+#define CS42L84_MSM_BLOCK_EN3 0x1802
+#define CS42L84_MSM_BLOCK_EN3_TR_SENSE BIT(3)
+
+#define CS42L84_HS_DET_CTL2 0x1811
+#define CS42L84_HS_DET_CTL2_CTL GENMASK(7, 6)
+#define CS42L84_HS_DET_CTL2_SET GENMASK(5, 4)
+#define CS42L84_HS_DET_CTL2_REF BIT(3)
+#define CS42L84_HS_DET_CTL2_AUTO_TIME GENMASK(1, 0)
+
+#define CS42L84_HS_SWITCH_CTL 0x1812
+#define CS42L84_HS_SWITCH_CTL_REF_HS3 BIT(7)
+#define CS42L84_HS_SWITCH_CTL_REF_HS4 BIT(6)
+#define CS42L84_HS_SWITCH_CTL_HSB_FILT_HS3 BIT(5)
+#define CS42L84_HS_SWITCH_CTL_HSB_FILT_HS4 BIT(4)
+#define CS42L84_HS_SWITCH_CTL_HSB_HS3 BIT(3)
+#define CS42L84_HS_SWITCH_CTL_HSB_HS4 BIT(2)
+#define CS42L84_HS_SWITCH_CTL_GNDHS_HS3 BIT(1)
+#define CS42L84_HS_SWITCH_CTL_GNDHS_HS4 BIT(0)
+
+#define CS42L84_HS_CLAMP_DISABLE 0x1813
+
+#define CS42L84_ADC_CTL1 0x2000
+#define CS42L84_ADC_CTL1_PREAMP_GAIN_SHIFT 6
+#define CS42L84_ADC_CTL1_PGA_GAIN_SHIFT 0
+#define CS42L84_ADC_CTL4 0x2003
+#define CS42L84_ADC_CTL4_WNF_CF_SHIFT 4
+#define CS42L84_ADC_CTL4_WNF_EN_SHIFT 3
+#define CS42L84_ADC_CTL4_HPF_CF_SHIFT 1
+#define CS42L84_ADC_CTL4_HPF_EN_SHIFT 0
+
+#define CS42L84_DAC_CTL1 0x3000
+#define CS42L84_DAC_CTL1_UNMUTE BIT(0)
+//#define CS42L84_DAC_CTL1_DACB_INV_SHIFT 1
+//#define CS42L84_DAC_CTL1_DACA_INV_SHIFT 0
+#define CS42L84_DAC_CTL2 0x3001
+
+#define CS42L84_DAC_CHA_VOL_LSB 0x3004
+#define CS42L84_DAC_CHA_VOL_MSB 0x3005
+#define CS42L84_DAC_CHB_VOL_LSB 0x3006
+#define CS42L84_DAC_CHB_VOL_MSB 0x3007
+#define CS42L84_HP_VOL_CTL 0x3020
+#define CS42L84_HP_VOL_CTL_ZERO_CROSS BIT(1)
+#define CS42L84_HP_VOL_CTL_SOFT BIT(0)
+
+#define CS42L84_SRC_ASP_RX_CH1 0b1101
+#define CS42L84_SRC_ASP_RX_CH2 0b1110
+
+#define CS42L84_BUS_ASP_TX_SRC 0x4000
+#define CS42L84_BUS_ASP_TX_SRC_CH1_SHIFT 0
+#define CS42L84_BUS_DAC_SRC 0x4001
+#define CS42L84_BUS_DAC_SRC_DACA_SHIFT 0
+#define CS42L84_BUS_DAC_SRC_DACB_SHIFT 4
+
+#define CS42L84_ASP_CTL 0x5000
+#define CS42L84_ASP_CTL_BCLK_EN_SHIFT 1
+#define CS42L84_ASP_CTL_TDM_MODE BIT(2)
+#define CS42L84_ASP_FSYNC_CTL2 0x5010
+#define CS42L84_ASP_FSYNC_CTL2_BCLK_PERIOD_LO GENMASK(7, 1)
+#define CS42L84_ASP_FSYNC_CTL3 0x5011
+#define CS42L84_ASP_FSYNC_CTL3_BCLK_PERIOD_HI GENMASK(4, 0)
+#define CS42L84_ASP_DATA_CTL 0x5018
+
+#define CS42L84_ASP_RX_EN 0x5020
+#define CS42L84_ASP_RX_EN_CH1_SHIFT 0
+#define CS42L84_ASP_RX_EN_CH2_SHIFT 1
+#define CS42L84_ASP_TX_EN 0x5024
+#define CS42L84_ASP_TX_EN_CH1_SHIFT 0
+
+#define CS42L84_ASP_RX_CH1_CTL1 0x5028
+#define CS42L84_ASP_RX_CH1_CTL2 0x5029
+#define CS42L84_ASP_RX_CH1_WIDTH 0x502a
+#define CS42L84_ASP_RX_CH2_CTL1 0x502c
+#define CS42L84_ASP_RX_CH2_CTL2 0x502d
+#define CS42L84_ASP_RX_CH2_WIDTH 0x502e
+
+#define CS42L84_ASP_RX_CHx_CTL1_EDGE BIT(0)
+#define CS42L84_ASP_RX_CHx_CTL1_SLOT_START_LSB GENMASK(7, 1)
+#define CS42L84_ASP_RX_CHx_CTL2_SLOT_START_MSB GENMASK(2, 0)
+
+#define CS42L84_ASP_TX_CH1_CTL1 0x5068
+#define CS42L84_ASP_TX_CH1_CTL2 0x5069
+#define CS42L84_ASP_TX_CH1_WIDTH 0x506a
+#define CS42L84_ASP_TX_CH2_CTL1 0x506c
+#define CS42L84_ASP_TX_CH2_CTL2 0x506d
+#define CS42L84_ASP_TX_CH2_WIDTH 0x506e
+
+#define CS42L84_DEBOUNCE_TIME_125MS 0b001
+#define CS42L84_DEBOUNCE_TIME_500MS 0b011
+
+#define CS42L84_BOOT_TIME_US 3000
+#define CS42L84_CLOCK_SWITCH_DELAY_US 150
+#define CS42L84_PLL_LOCK_POLL_US 250
+#define CS42L84_PLL_LOCK_TIMEOUT_US 1250
+
+#endif /* __CS42L84_H__ */
diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index 51b87a936179..c0d722de808f 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -16,6 +16,7 @@
#include <linux/regmap.h>
#include <linux/of.h>
#include <linux/of_gpio.h>
+#include <linux/of_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <sound/pcm.h>
@@ -25,6 +26,11 @@
#include "tas2764.h"
+enum tas2764_devid {
+ DEVID_TAS2764 = 0,
+ DEVID_SN012776 = 1
+};
+
struct tas2764_priv {
struct snd_soc_component *component;
struct gpio_desc *reset_gpio;
@@ -32,6 +38,7 @@ struct tas2764_priv {
struct regmap *regmap;
struct device *dev;
int irq;
+ enum tas2764_devid devid;
int v_sense_slot;
int i_sense_slot;
@@ -438,20 +445,13 @@ static int tas2764_set_dai_tdm_slot(struct snd_soc_dai *dai,
if (tx_mask == 0 || rx_mask != 0)
return -EINVAL;
- if (slots == 1) {
- if (tx_mask != 1)
- return -EINVAL;
- left_slot = 0;
- right_slot = 0;
+ left_slot = __ffs(tx_mask);
+ tx_mask &= ~(1 << left_slot);
+ if (tx_mask == 0) {
+ right_slot = left_slot;
} else {
- left_slot = __ffs(tx_mask);
- tx_mask &= ~(1 << left_slot);
- if (tx_mask == 0) {
- right_slot = left_slot;
- } else {
- right_slot = __ffs(tx_mask);
- tx_mask &= ~(1 << right_slot);
- }
+ right_slot = __ffs(tx_mask);
+ tx_mask &= ~(1 << right_slot);
}
if (tx_mask != 0 || left_slot >= slots || right_slot >= slots)
@@ -535,10 +535,16 @@ static struct snd_soc_dai_driver tas2764_dai_driver[] = {
},
};
+static uint8_t sn012776_bop_presets[] = {
+ 0x01, 0x32, 0x02, 0x22, 0x83, 0x2d, 0x80, 0x02, 0x06,
+ 0x32, 0x46, 0x30, 0x02, 0x06, 0x38, 0x40, 0x30, 0x02,
+ 0x06, 0x3e, 0x37, 0x30, 0xff, 0xe6
+};
+
static int tas2764_codec_probe(struct snd_soc_component *component)
{
struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
- int ret;
+ int ret, i;
tas2764->component = component;
@@ -587,6 +593,23 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
if (ret < 0)
return ret;
+ if (tas2764->devid == DEVID_SN012776) {
+ ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL,
+ TAS2764_PWR_CTRL_BOP_SRC,
+ TAS2764_PWR_CTRL_BOP_SRC);
+ if (ret < 0)
+ return ret;
+
+ for (i = 0; i < ARRAY_SIZE(sn012776_bop_presets); i++) {
+ ret = snd_soc_component_write(component,
+ TAS2764_BOP_CFG0 + i,
+ sn012776_bop_presets[i]);
+
+ if (ret < 0)
+ return ret;
+ }
+ }
+
return 0;
}
@@ -602,12 +625,21 @@ static SOC_ENUM_SINGLE_DECL(
tas2764_hpf_enum, TAS2764_DC_BLK0,
TAS2764_DC_BLK0_HPF_FREQ_PB_SHIFT, tas2764_hpf_texts);
+static const char * const tas2764_oce_texts[] = {
+ "Disable", "Retry",
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ tas2764_oce_enum, TAS2764_MISC_CFG1,
+ TAS2764_MISC_CFG1_OCE_RETRY_SHIFT, tas2764_oce_texts);
+
static const struct snd_kcontrol_new tas2764_snd_controls[] = {
SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0,
TAS2764_DVC_MAX, 1, tas2764_playback_volume),
SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0,
tas2764_digital_tlv),
SOC_ENUM("HPF Corner Frequency", tas2764_hpf_enum),
+ SOC_ENUM("OCE Handling", tas2764_oce_enum),
};
static const struct snd_soc_component_driver soc_component_driver_tas2764 = {
@@ -706,9 +738,12 @@ static int tas2764_parse_dt(struct device *dev, struct tas2764_priv *tas2764)
return 0;
}
+static const struct of_device_id tas2764_of_match[];
+
static int tas2764_i2c_probe(struct i2c_client *client)
{
struct tas2764_priv *tas2764;
+ const struct of_device_id *of_id = NULL;
int result;
tas2764 = devm_kzalloc(&client->dev, sizeof(struct tas2764_priv),
@@ -716,6 +751,14 @@ static int tas2764_i2c_probe(struct i2c_client *client)
if (!tas2764)
return -ENOMEM;
+ if (client->dev.of_node)
+ of_id = of_match_device(tas2764_of_match, &client->dev);
+
+ if (of_id)
+ tas2764->devid = (enum tas2764_devid) of_id->data;
+ else
+ tas2764->devid = DEVID_TAS2764;
+
tas2764->dev = &client->dev;
tas2764->irq = client->irq;
i2c_set_clientdata(client, tas2764);
@@ -752,7 +795,8 @@ MODULE_DEVICE_TABLE(i2c, tas2764_i2c_id);
#if defined(CONFIG_OF)
static const struct of_device_id tas2764_of_match[] = {
- { .compatible = "ti,tas2764" },
+ { .compatible = "ti,tas2764", .data = (void*) DEVID_TAS2764 },
+ { .compatible = "ti,sn012776", .data = (void*) DEVID_SN012776 },
{},
};
MODULE_DEVICE_TABLE(of, tas2764_of_match);
diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h
index 168af772a898..20628e51bf94 100644
--- a/sound/soc/codecs/tas2764.h
+++ b/sound/soc/codecs/tas2764.h
@@ -29,6 +29,7 @@
#define TAS2764_PWR_CTRL_ACTIVE 0x0
#define TAS2764_PWR_CTRL_MUTE BIT(0)
#define TAS2764_PWR_CTRL_SHUTDOWN BIT(1)
+#define TAS2764_PWR_CTRL_BOP_SRC BIT(7)
#define TAS2764_VSENSE_POWER_EN 3
#define TAS2764_ISENSE_POWER_EN 4
@@ -43,6 +44,10 @@
#define TAS2764_CHNL_0 TAS2764_REG(0X0, 0x03)
+/* Miscellaneous */
+#define TAS2764_MISC_CFG1 TAS2764_REG(0x0, 0x06)
+#define TAS2764_MISC_CFG1_OCE_RETRY_SHIFT 5
+
/* TDM Configuration Reg0 */
#define TAS2764_TDM_CFG0 TAS2764_REG(0X0, 0x08)
#define TAS2764_TDM_CFG0_SMP_MASK BIT(5)
@@ -110,4 +115,6 @@
#define TAS2764_INT_CLK_CFG TAS2764_REG(0x0, 0x5c)
#define TAS2764_INT_CLK_CFG_IRQZ_CLR BIT(2)
+#define TAS2764_BOP_CFG0 TAS2764_REG(0X0, 0x1d)
+
#endif /* __TAS2764__ */
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index b6765235a4b3..8557759acb1f 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -395,21 +395,13 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai,
if (tx_mask == 0 || rx_mask != 0)
return -EINVAL;
- if (slots == 1) {
- if (tx_mask != 1)
- return -EINVAL;
-
- left_slot = 0;
- right_slot = 0;
+ left_slot = __ffs(tx_mask);
+ tx_mask &= ~(1 << left_slot);
+ if (tx_mask == 0) {
+ right_slot = left_slot;
} else {
- left_slot = __ffs(tx_mask);
- tx_mask &= ~(1 << left_slot);
- if (tx_mask == 0) {
- right_slot = left_slot;
- } else {
- right_slot = __ffs(tx_mask);
- tx_mask &= ~(1 << right_slot);
- }
+ right_slot = __ffs(tx_mask);
+ tx_mask &= ~(1 << right_slot);
}
if (tx_mask != 0 || left_slot >= slots || right_slot >= slots)
diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c
index a6db6f0e5431..afdf0c863aa1 100644
--- a/sound/soc/codecs/tas2780.c
+++ b/sound/soc/codecs/tas2780.c
@@ -380,20 +380,13 @@ static int tas2780_set_dai_tdm_slot(struct snd_soc_dai *dai,
if (tx_mask == 0 || rx_mask != 0)
return -EINVAL;
- if (slots == 1) {
- if (tx_mask != 1)
- return -EINVAL;
- left_slot = 0;
- right_slot = 0;
+ left_slot = __ffs(tx_mask);
+ tx_mask &= ~(1 << left_slot);
+ if (tx_mask == 0) {
+ right_slot = left_slot;
} else {
- left_slot = __ffs(tx_mask);
- tx_mask &= ~(1 << left_slot);
- if (tx_mask == 0) {
- right_slot = left_slot;
- } else {
- right_slot = __ffs(tx_mask);
- tx_mask &= ~(1 << right_slot);
- }
+ right_slot = __ffs(tx_mask);
+ tx_mask &= ~(1 << right_slot);
}
if (tx_mask != 0 || left_slot >= slots || right_slot >= slots)
diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c
index 285ab4c9c716..674a12258fc7 100644
--- a/sound/soc/soc-card.c
+++ b/sound/soc/soc-card.c
@@ -197,10 +197,16 @@ int snd_soc_card_late_probe(struct snd_soc_card *card)
return 0;
}
-void snd_soc_card_fixup_controls(struct snd_soc_card *card)
+int snd_soc_card_fixup_controls(struct snd_soc_card *card)
{
- if (card->fixup_controls)
- card->fixup_controls(card);
+ if (card->fixup_controls) {
+ int ret = card->fixup_controls(card);
+
+ if (ret < 0)
+ return soc_card_ret(card, ret);
+ }
+
+ return 0;
}
int snd_soc_card_remove(struct snd_soc_card *card)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 12a82f5a3ff6..a6a7a9118271 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2071,7 +2071,10 @@ static int snd_soc_bind_card(struct snd_soc_card *card)
goto probe_end;
snd_soc_dapm_new_widgets(card);
- snd_soc_card_fixup_controls(card);
+
+ ret = snd_soc_card_fixup_controls(card);
+ if (ret < 0)
+ goto probe_end;
ret = snd_card_register(card->snd_card);
if (ret < 0) {
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d515e7a78ea8..248d5f7d86f9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2210,6 +2210,139 @@ static const struct file_operations dapm_bias_fops = {
.llseek = default_llseek,
};
+static ssize_t dapm_graph_read_file(struct file *file, char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ struct snd_soc_card *card = file->private_data;
+ struct snd_soc_dapm_context *dapm;
+ struct snd_soc_dapm_path *p;
+ struct snd_soc_dapm_widget *w;
+ struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dapm_widget *wdone[16];
+ struct snd_soc_dai *dai;
+ int i, num_wdone = 0, cluster = 0;
+ char *buf;
+ ssize_t bufsize;
+ ssize_t ret = 0;
+
+ bufsize = 1024 * card->num_dapm_widgets;
+ buf = kmalloc(bufsize, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ mutex_lock(&card->dapm_mutex);
+
+#define bufprintf(...) \
+ ret += scnprintf(buf + ret, bufsize - ret, __VA_ARGS__)
+
+ bufprintf("digraph dapm {\n");
+
+ /*
+ * Print the user-visible devices of the card.
+ */
+ bufprintf("subgraph cluster_%d {\n", cluster++);
+ bufprintf("label=\"Devices\";style=filled;fillcolor=gray;\n");
+ for_each_card_rtds(card, rtd) {
+ if (rtd->dai_link->no_pcm)
+ continue;
+
+ bufprintf("w%pK [label=\"%d: %s\"];\n", rtd,
+ rtd->pcm->device, rtd->dai_link->name);
+ }
+ bufprintf("};\n");
+
+ /*
+ * Print the playback/capture widgets of DAIs just next to
+ * the user-visible devices. Keep the list of already printed
+ * widgets in 'wdone', so they will be skipped later.
+ */
+ for_each_card_rtds(card, rtd) {
+ for_each_rtd_cpu_dais(rtd, i, dai) {
+ if (dai->playback_widget) {
+ w = dai->playback_widget;
+ bufprintf("w%pK [label=\"%s\"];\n", w, w->name);
+ if (!rtd->dai_link->no_pcm)
+ bufprintf("w%pK -> w%pK;\n", rtd, w);
+ wdone[num_wdone] = w;
+ if (num_wdone < ARRAY_SIZE(wdone))
+ num_wdone++;
+ }
+
+ if (dai->capture_widget) {
+ w = dai->capture_widget;
+ bufprintf("w%pK [label=\"%s\"];\n", w, w->name);
+ if (!rtd->dai_link->no_pcm)
+ bufprintf("w%pK -> w%pK;\n", w, rtd);
+ wdone[num_wdone] = w;
+ if (num_wdone < ARRAY_SIZE(wdone))
+ num_wdone++;
+ }
+ }
+ }
+
+ for_each_card_dapms(card, dapm) {
+ const char *prefix = soc_dapm_prefix(dapm);
+
+ if (dapm != &card->dapm) {
+ bufprintf("subgraph cluster_%d {\n", cluster++);
+ if (prefix)
+ bufprintf("label=\"%s\";\n", prefix);
+ else if (dapm->component)
+ bufprintf("label=\"%s\";\n",
+ dapm->component->name);
+ }
+
+ for_each_card_widgets(dapm->card, w) {
+ const char *name = w->name;
+ bool skip = false;
+
+ if (w->dapm != dapm)
+ continue;
+
+ if (list_empty(&w->edges[0]) && list_empty(&w->edges[1]))
+ continue;
+
+ for (i = 0; i < num_wdone; i++)
+ if (wdone[i] == w)
+ skip = true;
+ if (skip)
+ continue;
+
+ if (prefix && strlen(name) > strlen(prefix) + 1)
+ name += strlen(prefix) + 1;
+
+ bufprintf("w%pK [label=\"%s\"];\n", w, name);
+ }
+
+ if (dapm != &card->dapm)
+ bufprintf("}\n");
+ }
+
+ list_for_each_entry(p, &card->paths, list) {
+ if (p->name)
+ bufprintf("w%pK -> w%pK [label=\"%s\"];\n",
+ p->source, p->sink, p->name);
+ else
+ bufprintf("w%pK -> w%pK;\n", p->source, p->sink);
+ }
+
+ bufprintf("}\n");
+#undef bufprintf
+
+ mutex_unlock(&card->dapm_mutex);
+
+ ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+
+ kfree(buf);
+ return ret;
+}
+
+static const struct file_operations dapm_graph_fops = {
+ .open = simple_open,
+ .read = dapm_graph_read_file,
+ .llseek = default_llseek,
+};
+
void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
struct dentry *parent)
{
@@ -2220,6 +2353,10 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm,
&dapm_bias_fops);
+
+ if (dapm == &dapm->card->dapm)
+ debugfs_create_file("graph.dot", 0444, dapm->debugfs_dapm,
+ dapm->card, &dapm_graph_fops);
}
static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index bd88de056358..2eb8a370eaaf 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -176,28 +176,20 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
- const char *vol_string = NULL;
- int max;
+ int platform_max;
- max = uinfo->value.integer.max = mc->max - mc->min;
- if (mc->platform_max && mc->platform_max < max)
- max = mc->platform_max;
+ if (!mc->platform_max)
+ mc->platform_max = mc->max;
+ platform_max = mc->platform_max;
- if (max == 1) {
- /* Even two value controls ending in Volume should always be integer */
- vol_string = strstr(kcontrol->id.name, " Volume");
- if (vol_string && !strcmp(vol_string, " Volume"))
- uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- else
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- } else {
+ if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume"))
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- }
uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = max;
-
+ uinfo->value.integer.max = platform_max - mc->min;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
@@ -634,37 +626,217 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
+static bool soc_control_matches(struct snd_kcontrol *kctl,
+ const char *pattern)
+{
+ const char *name = kctl->id.name;
+
+ if (pattern[0] == '*') {
+ int namelen;
+ int patternlen;
+
+ pattern++;
+ if (pattern[0] == ' ')
+ pattern++;
+
+ namelen = strlen(name);
+ patternlen = strlen(pattern);
+
+ if (namelen > patternlen)
+ name += namelen - patternlen;
+ }
+
+ return !strcmp(name, pattern);
+}
+
+static int soc_clip_to_platform_max(struct snd_kcontrol *kctl)
+{
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)kctl->private_value;
+ struct snd_ctl_elem_value uctl;
+ int ret;
+
+ if (!mc->platform_max)
+ return 0;
+
+ ret = kctl->get(kctl, &uctl);
+ if (ret < 0)
+ return ret;
+
+ if (uctl.value.integer.value[0] > mc->platform_max)
+ uctl.value.integer.value[0] = mc->platform_max;
+
+ if (snd_soc_volsw_is_stereo(mc) &&
+ uctl.value.integer.value[1] > mc->platform_max)
+ uctl.value.integer.value[1] = mc->platform_max;
+
+ ret = kctl->put(kctl, &uctl);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int soc_limit_volume(struct snd_kcontrol *kctl, int max)
+{
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)kctl->private_value;
+
+ if (max <= 0 || max > mc->max)
+ return -EINVAL;
+ mc->platform_max = max;
+
+ return soc_clip_to_platform_max(kctl);
+}
+
/**
- * snd_soc_limit_volume - Set new limit to an existing volume control.
+ * snd_soc_limit_volume - Set new limit to existing volume controls
*
* @card: where to look for the control
- * @name: Name of the control
+ * @name: name pattern
* @max: new maximum limit
+ *
+ * Finds controls matching the given name (which can be either a name
+ * verbatim, or a pattern starting with the wildcard '*') and sets
+ * a platform volume limit on them.
*
- * Return 0 for success, else error.
+ * Return number of matching controls on success, else error. At least
+ * one control needs to match the pattern.
*/
int snd_soc_limit_volume(struct snd_soc_card *card,
const char *name, int max)
{
struct snd_kcontrol *kctl;
- int ret = -EINVAL;
+ int hits = 0;
+ int ret;
- /* Sanity check for name and max */
- if (unlikely(!name || max <= 0))
+ /* Sanity check for name */
+ if (unlikely(!name))
return -EINVAL;
- kctl = snd_soc_card_get_kcontrol(card, name);
- if (kctl) {
- struct soc_mixer_control *mc = (struct soc_mixer_control *)kctl->private_value;
- if (max <= mc->max) {
- mc->platform_max = max;
- ret = 0;
- }
+ list_for_each_entry(kctl, &card->snd_card->controls, list) {
+ if (!soc_control_matches(kctl, name))
+ continue;
+
+ ret = soc_limit_volume(kctl, max);
+ if (ret < 0)
+ return ret;
+ hits++;
}
- return ret;
+
+ if (!hits)
+ return -EINVAL;
+
+ return hits;
}
EXPORT_SYMBOL_GPL(snd_soc_limit_volume);
+/**
+ * snd_soc_deactivate_kctl - Activate/deactive controls matching a pattern
+ *
+ * @card: where to look for the controls
+ * @name: name pattern
+ * @active: non-zero to activate, zero to deactivate
+ *
+ * Return number of matching controls on success, else error.
+ * No controls need to match.
+ */
+int snd_soc_deactivate_kctl(struct snd_soc_card *card,
+ const char *name, int active)
+{
+ struct snd_kcontrol *kctl;
+ int hits = 0;
+ int ret;
+
+ /* Sanity check for name */
+ if (unlikely(!name))
+ return -EINVAL;
+
+ list_for_each_entry(kctl, &card->snd_card->controls, list) {
+ if (!soc_control_matches(kctl, name))
+ continue;
+
+ ret = snd_ctl_activate_id(card->snd_card, &kctl->id, active);
+ if (ret < 0)
+ return ret;
+ hits++;
+ }
+
+ if (!hits)
+ return -EINVAL;
+
+ return hits;
+}
+EXPORT_SYMBOL_GPL(snd_soc_deactivate_kctl);
+
+static int soc_set_enum_kctl(struct snd_kcontrol *kctl, const char *strval)
+{
+ struct snd_ctl_elem_value value;
+ struct snd_ctl_elem_info info;
+ int sel, i, ret;
+
+ ret = kctl->info(kctl, &info);
+ if (ret < 0)
+ return ret;
+
+ if (info.type != SNDRV_CTL_ELEM_TYPE_ENUMERATED)
+ return -EINVAL;
+
+ for (sel = 0; sel < info.value.enumerated.items; sel++) {
+ info.value.enumerated.item = sel;
+ ret = kctl->info(kctl, &info);
+ if (ret < 0)
+ return ret;
+
+ if (!strcmp(strval, info.value.enumerated.name))
+ break;
+ }
+
+ if (sel == info.value.enumerated.items)
+ return -EINVAL;
+
+ for (i = 0; i < info.count; i++)
+ value.value.enumerated.item[i] = sel;
+
+ return kctl->put(kctl, &value);
+}
+
+/**
+ * snd_soc_set_enum_kctl - Set enumerated controls matching a pattern
+ *
+ * @card: where to look for the controls
+ * @name: name pattern
+ * @value: string value to set the controls to
+ *
+ * Return number of matching and set controls on success, else error.
+ * No controls need to match.
+ */
+int snd_soc_set_enum_kctl(struct snd_soc_card *card,
+ const char *name, const char *value)
+{
+ struct snd_kcontrol *kctl;
+ int hits = 0;
+ int ret;
+
+ /* Sanity check for name */
+ if (unlikely(!name))
+ return -EINVAL;
+
+ list_for_each_entry(kctl, &card->snd_card->controls, list) {
+ if (!soc_control_matches(kctl, name))
+ continue;
+
+ ret = soc_set_enum_kctl(kctl, value);
+ if (ret < 0)
+ return ret;
+ hits++;
+ }
+
+ if (!hits)
+ return -EINVAL;
+
+ return hits;
+}
+EXPORT_SYMBOL_GPL(snd_soc_set_enum_kctl);
+
int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-22 04:53:52 +0000