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Diffstat (limited to 'sound/soc/fsl/fsl-asoc-card.c')
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c216
1 files changed, 180 insertions, 36 deletions
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 00be73900888..de136c0a497d 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -15,6 +15,8 @@
#endif
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/simple_card_utils.h>
#include "fsl_esai.h"
#include "fsl_sai.h"
@@ -33,8 +35,7 @@
#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
/**
- * CODEC private data
- *
+ * struct codec_priv - CODEC private data
* @mclk_freq: Clock rate of MCLK
* @mclk_id: MCLK (or main clock) id for set_sysclk()
* @fll_id: FLL (or secordary clock) id for set_sysclk()
@@ -48,11 +49,10 @@ struct codec_priv {
};
/**
- * CPU private data
- *
- * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
- * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
- * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ * struct cpu_priv - CPU private data
+ * @sysclk_freq: SYSCLK rates for set_sysclk()
+ * @sysclk_dir: SYSCLK directions for set_sysclk()
+ * @sysclk_id: SYSCLK ids for set_sysclk()
* @slot_width: Slot width of each frame
*
* Note: [1] for tx and [0] for rx
@@ -65,9 +65,10 @@ struct cpu_priv {
};
/**
- * Freescale Generic ASOC card private data
- *
- * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
+ * @dai_link: DAI link structure including normal one and DPCM link
+ * @hp_jack: Headphone Jack structure
+ * @mic_jack: Microphone Jack structure
* @pdev: platform device pointer
* @codec_priv: CODEC private data
* @cpu_priv: CPU private data
@@ -82,6 +83,8 @@ struct cpu_priv {
struct fsl_asoc_card_priv {
struct snd_soc_dai_link dai_link[3];
+ struct asoc_simple_jack hp_jack;
+ struct asoc_simple_jack mic_jack;
struct platform_device *pdev;
struct codec_priv codec_priv;
struct cpu_priv cpu_priv;
@@ -94,8 +97,8 @@ struct fsl_asoc_card_priv {
char name[32];
};
-/**
- * This dapm route map exsits for DPCM link only.
+/*
+ * This dapm route map exists for DPCM link only.
* The other routes shall go through Device Tree.
*
* Note: keep all ASRC routes in the second half
@@ -119,6 +122,13 @@ static const struct snd_soc_dapm_route audio_map_ac97[] = {
{"ASRC-Capture", NULL, "AC97 Capture"},
};
+static const struct snd_soc_dapm_route audio_map_tx[] = {
+ /* 1st half -- Normal DAPM routes */
+ {"Playback", NULL, "CPU-Playback"},
+ /* 2nd half -- ASRC DAPM routes */
+ {"CPU-Playback", NULL, "ASRC-Playback"},
+};
+
/* Add all possible widgets into here without being redundant */
static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
SND_SOC_DAPM_LINE("Line Out Jack", NULL),
@@ -138,7 +148,7 @@ static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
struct cpu_priv *cpu_priv = &priv->cpu_priv;
@@ -441,6 +451,44 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
return 0;
}
+static int hp_jack_event(struct notifier_block *nb, unsigned long event,
+ void *data)
+{
+ struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+ struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+ if (event & SND_JACK_HEADPHONE)
+ /* Disable speaker if headphone is plugged in */
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ else
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+
+ return 0;
+}
+
+static struct notifier_block hp_jack_nb = {
+ .notifier_call = hp_jack_event,
+};
+
+static int mic_jack_event(struct notifier_block *nb, unsigned long event,
+ void *data)
+{
+ struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+ struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+ if (event & SND_JACK_MICROPHONE)
+ /* Disable dmic if microphone is plugged in */
+ snd_soc_dapm_disable_pin(dapm, "DMIC");
+ else
+ snd_soc_dapm_enable_pin(dapm, "DMIC");
+
+ return 0;
+}
+
+static struct notifier_block mic_jack_nb = {
+ .notifier_call = mic_jack_event,
+};
+
static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
@@ -483,10 +531,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
struct device_node *cpu_np, *codec_np, *asrc_np;
struct device_node *np = pdev->dev.of_node;
struct platform_device *asrc_pdev = NULL;
+ struct device_node *bitclkmaster = NULL;
+ struct device_node *framemaster = NULL;
struct platform_device *cpu_pdev;
struct fsl_asoc_card_priv *priv;
- struct i2c_client *codec_dev;
+ struct device *codec_dev = NULL;
const char *codec_dai_name;
+ const char *codec_dev_name;
+ unsigned int daifmt;
u32 width;
int ret;
@@ -512,10 +564,23 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
}
codec_np = of_parse_phandle(np, "audio-codec", 0);
- if (codec_np)
- codec_dev = of_find_i2c_device_by_node(codec_np);
- else
- codec_dev = NULL;
+ if (codec_np) {
+ struct platform_device *codec_pdev;
+ struct i2c_client *codec_i2c;
+
+ codec_i2c = of_find_i2c_device_by_node(codec_np);
+ if (codec_i2c) {
+ codec_dev = &codec_i2c->dev;
+ codec_dev_name = codec_i2c->name;
+ }
+ if (!codec_dev) {
+ codec_pdev = of_find_device_by_node(codec_np);
+ if (codec_pdev) {
+ codec_dev = &codec_pdev->dev;
+ codec_dev_name = codec_pdev->name;
+ }
+ }
+ }
asrc_np = of_parse_phandle(np, "audio-asrc", 0);
if (asrc_np)
@@ -523,7 +588,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
if (codec_dev) {
- struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
+ struct clk *codec_clk = clk_get(codec_dev, NULL);
if (!IS_ERR(codec_clk)) {
priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
@@ -538,6 +603,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
/* Assign a default DAI format, and allow each card to overwrite it */
priv->dai_fmt = DAI_FMT_BASE;
+ memcpy(priv->dai_link, fsl_asoc_card_dai,
+ sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+ priv->card.dapm_routes = audio_map;
+ priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
/* Diversify the card configurations */
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
codec_dai_name = "cs42888";
@@ -573,12 +643,58 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
codec_dai_name = "ac97-hifi";
priv->card.set_bias_level = NULL;
priv->dai_fmt = SND_SOC_DAIFMT_AC97;
+ priv->card.dapm_routes = audio_map_ac97;
+ priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
+ } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
+ codec_dai_name = "fsl-mqs-dai";
+ priv->card.set_bias_level = NULL;
+ priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
+ SND_SOC_DAIFMT_CBS_CFS |
+ SND_SOC_DAIFMT_NB_NF;
+ priv->dai_link[1].dpcm_capture = 0;
+ priv->dai_link[2].dpcm_capture = 0;
+ priv->card.dapm_routes = audio_map_tx;
+ priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
+ } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
+ codec_dai_name = "wm8524-hifi";
+ priv->card.set_bias_level = NULL;
+ priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+ priv->dai_link[1].dpcm_capture = 0;
+ priv->dai_link[2].dpcm_capture = 0;
+ priv->cpu_priv.slot_width = 32;
+ priv->card.dapm_routes = audio_map_tx;
+ priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
goto asrc_fail;
}
+ /* Format info from DT is optional. */
+ daifmt = snd_soc_of_parse_daifmt(np, NULL,
+ &bitclkmaster, &framemaster);
+ daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+ if (bitclkmaster || framemaster) {
+ if (codec_np == bitclkmaster)
+ daifmt |= (codec_np == framemaster) ?
+ SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
+ else
+ daifmt |= (codec_np == framemaster) ?
+ SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
+
+ /* Override dai_fmt with value from DT */
+ priv->dai_fmt = daifmt;
+ }
+
+ /* Change direction according to format */
+ if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) {
+ priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
+ priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
+ }
+
+ of_node_put(bitclkmaster);
+ of_node_put(framemaster);
+
if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
dev_err(&pdev->dev, "failed to find codec device\n");
ret = -EPROBE_DEFER;
@@ -601,19 +717,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
}
- snprintf(priv->name, sizeof(priv->name), "%s-audio",
- fsl_asoc_card_is_ac97(priv) ? "ac97" :
- codec_dev->name);
-
/* Initialize sound card */
priv->pdev = pdev;
priv->card.dev = &pdev->dev;
- priv->card.name = priv->name;
+ ret = snd_soc_of_parse_card_name(&priv->card, "model");
+ if (ret) {
+ snprintf(priv->name, sizeof(priv->name), "%s-audio",
+ fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
+ priv->card.name = priv->name;
+ }
priv->card.dai_link = priv->dai_link;
- priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
- audio_map_ac97 : audio_map;
priv->card.late_probe = fsl_asoc_card_late_probe;
- priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
@@ -621,13 +735,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
if (!asrc_pdev)
priv->card.num_dapm_routes /= 2;
- memcpy(priv->dai_link, fsl_asoc_card_dai,
- sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
-
- ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
- if (ret) {
- dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
- goto asrc_fail;
+ if (of_property_read_bool(np, "audio-routing")) {
+ ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
+ if (ret) {
+ dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
+ goto asrc_fail;
+ }
}
/* Normal DAI Link */
@@ -704,8 +817,37 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(&priv->card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
- if (ret && ret != -EPROBE_DEFER)
- dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto asrc_fail;
+ }
+
+ /*
+ * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
+ * asoc_simple_init_jack uses these properties for creating
+ * Headphone Jack and Microphone Jack.
+ *
+ * The notifier is initialized in snd_soc_card_jack_new(), then
+ * snd_soc_jack_notifier_register can be called.
+ */
+ if (of_property_read_bool(np, "hp-det-gpio")) {
+ ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
+ 1, NULL, "Headphone Jack");
+ if (ret)
+ goto asrc_fail;
+
+ snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
+ }
+
+ if (of_property_read_bool(np, "mic-det-gpio")) {
+ ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
+ 0, NULL, "Mic Jack");
+ if (ret)
+ goto asrc_fail;
+
+ snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
+ }
asrc_fail:
of_node_put(asrc_np);
@@ -724,6 +866,8 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-sgtl5000", },
{ .compatible = "fsl,imx-audio-wm8962", },
{ .compatible = "fsl,imx-audio-wm8960", },
+ { .compatible = "fsl,imx-audio-mqs", },
+ { .compatible = "fsl,imx-audio-wm8524", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
Copyright © 1996 – 2023 Jason A. Donenfeld. All Rights Reverse Engineered.