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);