From ecbb1b8d97a78a72003fcb19292da502d393bf80 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 6 Jul 2017 00:39:29 +0000
Subject: ASoC: hdmi-codec: ELD control corresponds to the PCM stream

Current hdmi-codec driver is using hdmi_controls for "ELD" control.
But, hdmi-codec driver might be used from many HDMIs. Thus, we need
to correspond device number, otherwise we will receive below error.

        xxx: control x:x:x:ELD:x is already present

This patch registers ELD control in .pcm_new by using
.device = rtd->pcm->device to corresponding to PCM stream.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
[Takashi: use snd_ctl_new1()/snd_ctl_add()]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdmi-codec.c | 31 ++++++++++++++++---------------
 1 file changed, 16 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 22ed0dc88f0a..7686a80861f1 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -399,18 +399,6 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-
-static const struct snd_kcontrol_new hdmi_controls[] = {
-	{
-		.access = SNDRV_CTL_ELEM_ACCESS_READ |
-			  SNDRV_CTL_ELEM_ACCESS_VOLATILE,
-		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
-		.name = "ELD",
-		.info = hdmi_eld_ctl_info,
-		.get = hdmi_eld_ctl_get,
-	},
-};
-
 static int hdmi_codec_new_stream(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
@@ -668,6 +656,16 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_soc_dai_driver *drv = dai->driver;
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	struct snd_kcontrol *kctl;
+	struct snd_kcontrol_new hdmi_eld_ctl = {
+		.access	= SNDRV_CTL_ELEM_ACCESS_READ |
+			  SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.iface	= SNDRV_CTL_ELEM_IFACE_PCM,
+		.name	= "ELD",
+		.info	= hdmi_eld_ctl_info,
+		.get	= hdmi_eld_ctl_get,
+		.device	= rtd->pcm->device,
+	};
 	int ret;
 
 	dev_dbg(dai->dev, "%s()\n", __func__);
@@ -686,7 +684,12 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 	hcp->chmap_info->chmap = hdmi_codec_stereo_chmaps;
 	hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN;
 
-	return 0;
+	/* add ELD ctl with the device number corresponding to the PCM stream */
+	kctl = snd_ctl_new1(&hdmi_eld_ctl, dai->component);
+	if (!kctl)
+		return -ENOMEM;
+
+	return snd_ctl_add(rtd->card->snd_card, kctl);
 }
 
 static struct snd_soc_dai_driver hdmi_i2s_dai = {
@@ -732,8 +735,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component,
 
 static struct snd_soc_codec_driver hdmi_codec = {
 	.component_driver = {
-		.controls		= hdmi_controls,
-		.num_controls		= ARRAY_SIZE(hdmi_controls),
 		.dapm_widgets		= hdmi_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(hdmi_widgets),
 		.dapm_routes		= hdmi_routes,
-- 
cgit v1.2.3


From 1d6463a3e23e55621101a1b9b842101988c596ff Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Tue, 11 Jul 2017 15:08:55 +0100
Subject: ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits
 static

Don't populate array hdmi_codec_eld_spk_alloc_bits on the stack but
make it static. Makes the object code smaller by over 260 bytes:

Before:
   text	   data	    bss	    dec	    hex	filename
  10882	   3384	     64	  14330	   37fa	sound/soc/codecs/hdmi-codec.o

After:
   text	   data	    bss	    dec	    hex	filename
  10557	   3440	     64	  14061	   36ed	sound/soc/codecs/hdmi-codec.o

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdmi-codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 7686a80861f1..509ab513b4b2 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -326,7 +326,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
 static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc)
 {
 	int i;
-	const unsigned long hdmi_codec_eld_spk_alloc_bits[] = {
+	static const unsigned long hdmi_codec_eld_spk_alloc_bits[] = {
 		[0] = FL | FR, [1] = LFE, [2] = FC, [3] = RL | RR,
 		[4] = RC, [5] = FLC | FRC, [6] = RLC | RRC,
 	};
-- 
cgit v1.2.3


From e7b2e30a3f88b02b13a9bde05044142b38456004 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Fri, 30 Jun 2017 16:46:33 -0500
Subject: ASoC: kirkwood-i2s: fix error return code in kirkwood_i2s_dev_probe()

platform_get_irq() returns an error code, but the kirkwood-i2s
driver ignores it and always returns -ENXIO. This is not correct,
and prevents -EPROBE_DEFER from being propagated properly.
Notice that platform_get_irq() no longer returns 0 on error.

Print and propagate the return value of platform_get_irq on failure.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/kirkwood/kirkwood-i2s.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3a36d60e1785..3b09f71b9535 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -550,9 +550,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return PTR_ERR(priv->io);
 
 	priv->irq = platform_get_irq(pdev, 0);
-	if (priv->irq <= 0) {
-		dev_err(&pdev->dev, "platform_get_irq failed\n");
-		return -ENXIO;
+	if (priv->irq < 0) {
+		dev_err(&pdev->dev, "platform_get_irq failed: %d\n", priv->irq);
+		return priv->irq;
 	}
 
 	if (np) {
-- 
cgit v1.2.3


From ea4daf81fc5a37766a606479f592cd9dfe6f1da8 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Mon, 10 Jul 2017 10:03:12 +0800
Subject: ASoC: rt5514: Move the auto disable DSP function to set_bias_level()

Move the auto disable DSP function from the widget event "DAPM Pre" to
the set_bias_level() function to avoid the unexpected behavior. While the
SPI was recording using the DAI "rt5514-dsp-cpu-dai", the event "DAPM Pre"
would be manipulated even the DAPM widgets were not connected, and it also
would power off the DSP that cause the DSP buffered data lost. So the auto
disable DSP function should move to the set_bias_level() function to make
sure that it is only manipulated by the DAPM widgets are connected.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 49 +++++++++++++++++------------------------------
 1 file changed, 18 insertions(+), 31 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 1b6796c4c471..a71e63097aaf 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -471,35 +471,6 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
 		return 0;
 }
 
-static int rt5514_pre_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-	struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec);
-
-	switch (event) {
-	case SND_SOC_DAPM_PRE_PMU:
-		/**
-		 * If the DSP is enabled in start of recording, the DSP
-		 * should be disabled, and sync back to normal recording
-		 * settings to make sure recording properly.
-		*/
-		if (rt5514->dsp_enabled) {
-			rt5514->dsp_enabled = 0;
-			regmap_multi_reg_write(rt5514->i2c_regmap,
-				rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch));
-			regcache_mark_dirty(rt5514->regmap);
-			regcache_sync(rt5514->regmap);
-		}
-		break;
-
-	default:
-		return 0;
-	}
-
-	return 0;
-}
-
 static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = {
 	/* Input Lines */
 	SND_SOC_DAPM_INPUT("DMIC1L"),
@@ -607,8 +578,6 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = {
 
 	/* Audio Interface */
 	SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
-
-	SND_SOC_DAPM_PRE("DAPM Pre", rt5514_pre_event),
 };
 
 static const struct snd_soc_dapm_route rt5514_dapm_routes[] = {
@@ -975,6 +944,24 @@ static int rt5514_set_bias_level(struct snd_soc_codec *codec,
 		}
 		break;
 
+	case SND_SOC_BIAS_STANDBY:
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+			/*
+			 * If the DSP is enabled in start of recording, the DSP
+			 * should be disabled, and sync back to normal recording
+			 * settings to make sure recording properly.
+			 */
+			if (rt5514->dsp_enabled) {
+				rt5514->dsp_enabled = 0;
+				regmap_multi_reg_write(rt5514->i2c_regmap,
+					rt5514_i2c_patch,
+					ARRAY_SIZE(rt5514_i2c_patch));
+				regcache_mark_dirty(rt5514->regmap);
+				regcache_sync(rt5514->regmap);
+			}
+		}
+		break;
+
 	default:
 		break;
 	}
-- 
cgit v1.2.3


From 7d8e00c78924d50eb71d3cf991096021eadce641 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Fri, 7 Jul 2017 16:58:57 +0800
Subject: ASoC: rt5663: Update the calibration funciton

The patch updates the calibration funciton for getting the correct DC
offset of headphone output.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 100 ++++++++++++++++++++++++++++++++++++----------
 1 file changed, 79 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index a33202affeb1..18b3b526b7c5 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -2986,47 +2986,75 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663)
 {
 	int value, count;
 
-	regmap_write(rt5663->regmap, RT5663_RC_CLK, 0x0280);
+	regmap_write(rt5663->regmap, RT5663_RESET, 0x0000);
+	msleep(20);
+	regmap_write(rt5663->regmap, RT5663_ANA_BIAS_CUR_4, 0x00a1);
+	regmap_write(rt5663->regmap, RT5663_RC_CLK, 0x0380);
 	regmap_write(rt5663->regmap, RT5663_GLB_CLK, 0x8000);
-	regmap_write(rt5663->regmap, RT5663_DIG_MISC, 0x8001);
+	regmap_write(rt5663->regmap, RT5663_ADDA_CLK_1, 0x1000);
 	regmap_write(rt5663->regmap, RT5663_VREF_RECMIX, 0x0032);
-	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa2be);
-	msleep(20);
-	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf2be);
-	regmap_write(rt5663->regmap, RT5663_PWR_DIG_2, 0x8400);
-	regmap_write(rt5663->regmap, RT5663_CHOP_ADC, 0x3000);
-	regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x003b);
-	regmap_write(rt5663->regmap, RT5663_PWR_DIG_1, 0x8df8);
-	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x0003);
-	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c);
-	regmap_write(rt5663->regmap, RT5663_ADDA_CLK_1, 0x1111);
+	regmap_write(rt5663->regmap, RT5663_HP_IMP_SEN_19, 0x000c);
+	regmap_write(rt5663->regmap, RT5663_DUMMY_1, 0x0324);
+	regmap_write(rt5663->regmap, RT5663_DIG_MISC, 0x8001);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa23b);
+	msleep(30);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf23b);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x8000);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x0008);
 	regmap_write(rt5663->regmap, RT5663_PRE_DIV_GATING_1, 0xffff);
 	regmap_write(rt5663->regmap, RT5663_PRE_DIV_GATING_2, 0xffff);
+	regmap_write(rt5663->regmap, RT5663_CBJ_1, 0x8c10);
+	regmap_write(rt5663->regmap, RT5663_IL_CMD_2, 0x00c1);
+	regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_1, 0xb880);
+	regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_2, 0x4110);
+	regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_2, 0x4118);
+
+	count = 0;
+	while (true) {
+		regmap_read(rt5663->regmap, RT5663_INT_ST_2, &value);
+		if (!(value & 0x80))
+			usleep_range(10000, 10005);
+		else
+			break;
+
+		if (count > 200)
+			break;
+	}
+
+	regmap_write(rt5663->regmap, RT5663_HP_IMP_SEN_19, 0x0000);
 	regmap_write(rt5663->regmap, RT5663_DEPOP_2, 0x3003);
+	regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x0038);
 	regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x003b);
+	regmap_write(rt5663->regmap, RT5663_PWR_DIG_2, 0x8400);
+	regmap_write(rt5663->regmap, RT5663_PWR_DIG_1, 0x8df8);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x8003);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c);
 	regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32);
-	regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_2, 0x1371);
 	regmap_write(rt5663->regmap, RT5663_DACREF_LDO, 0x3b0b);
-	regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x2080);
+	regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x0000);
 	regmap_write(rt5663->regmap, RT5663_BYPASS_STO_DAC, 0x000c);
-	regmap_write(rt5663->regmap, RT5663_HP_BIAS, 0xabba);
+	regmap_write(rt5663->regmap, RT5663_HP_BIAS, 0xafaa);
 	regmap_write(rt5663->regmap, RT5663_CHARGE_PUMP_1, 0x2224);
 	regmap_write(rt5663->regmap, RT5663_HP_OUT_EN, 0x8088);
 	regmap_write(rt5663->regmap, RT5663_STO_DRE_9, 0x0017);
 	regmap_write(rt5663->regmap, RT5663_STO_DRE_10, 0x0017);
 	regmap_write(rt5663->regmap, RT5663_STO1_ADC_MIXER, 0x4040);
+	regmap_write(rt5663->regmap, RT5663_CHOP_ADC, 0x3000);
 	regmap_write(rt5663->regmap, RT5663_RECMIX, 0x0005);
 	regmap_write(rt5663->regmap, RT5663_ADDA_RST, 0xc000);
 	regmap_write(rt5663->regmap, RT5663_STO1_HPF_ADJ1, 0x3320);
 	regmap_write(rt5663->regmap, RT5663_HP_CALIB_2, 0x00c9);
 	regmap_write(rt5663->regmap, RT5663_DUMMY_1, 0x004c);
-	regmap_write(rt5663->regmap, RT5663_ANA_BIAS_CUR_1, 0x7766);
-	regmap_write(rt5663->regmap, RT5663_BIAS_CUR_8, 0x4702);
-	msleep(200);
+	regmap_write(rt5663->regmap, RT5663_ANA_BIAS_CUR_1, 0x1111);
+	regmap_write(rt5663->regmap, RT5663_BIAS_CUR_8, 0x4402);
+	regmap_write(rt5663->regmap, RT5663_CHARGE_PUMP_2, 0x3311);
 	regmap_write(rt5663->regmap, RT5663_HP_CALIB_1, 0x0069);
-	regmap_write(rt5663->regmap, RT5663_HP_CALIB_3, 0x06c2);
-	regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0x7b00);
-	regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0xfb00);
+	regmap_write(rt5663->regmap, RT5663_HP_CALIB_3, 0x06ce);
+	regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0x6800);
+	regmap_write(rt5663->regmap, RT5663_CHARGE_PUMP_2, 0x1100);
+	regmap_write(rt5663->regmap, RT5663_HP_CALIB_7, 0x0057);
+	regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0xe800);
+
 	count = 0;
 	while (true) {
 		regmap_read(rt5663->regmap, RT5663_HP_CALIB_1_1, &value);
@@ -3039,6 +3067,36 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663)
 			return;
 		count++;
 	}
+
+	regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0x6200);
+	regmap_write(rt5663->regmap, RT5663_HP_CALIB_7, 0x0059);
+	regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0xe200);
+
+	count = 0;
+	while (true) {
+		regmap_read(rt5663->regmap, RT5663_HP_CALIB_1_1, &value);
+		if (value & 0x8000)
+			usleep_range(10000, 10005);
+		else
+			break;
+
+		if (count > 200)
+			return;
+		count++;
+	}
+
+	regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_1, 0xb8e0);
+	usleep_range(10000, 10005);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0x003b);
+	usleep_range(10000, 10005);
+	regmap_write(rt5663->regmap, RT5663_PWR_DIG_1, 0x0000);
+	usleep_range(10000, 10005);
+	regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x000b);
+	usleep_range(10000, 10005);
+	regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x0008);
+	usleep_range(10000, 10005);
+	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x0000);
+	usleep_range(10000, 10005);
 }
 
 static int rt5663_i2c_probe(struct i2c_client *i2c,
-- 
cgit v1.2.3


From 8f24412784bd0f544307bb184e3f264064f68ff0 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Fri, 7 Jul 2017 16:58:58 +0800
Subject: ASoC: rt5663: Optimize the Jack Type detection

The patch optimizes the Jack Type detection.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 23 ++++++++++++++---------
 1 file changed, 14 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 18b3b526b7c5..ebd9ae00e04d 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -1508,7 +1508,7 @@ static int rt5663_v2_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 {
 	struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec);
-	int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30};
+	int val, i = 0;
 
 	dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert);
 
@@ -1543,17 +1543,22 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			RT5663_IRQ_POW_SAV_MASK, RT5663_IRQ_POW_SAV_EN);
 		snd_soc_update_bits(codec, RT5663_IRQ_1,
 			RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN);
-		while (i < 5) {
-			msleep(sleep_time[i]);
-			val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) &
-				0x0003;
-			dev_dbg(codec->dev, "%s: MX-00e7 val=%x sleep %d\n",
-				__func__, val, sleep_time[i]);
-			i++;
-			if (val == 0x1 || val == 0x2 || val == 0x3)
+
+		while (true) {
+			regmap_read(rt5663->regmap, RT5663_INT_ST_2, &val);
+			if (!(val & 0x80))
+				usleep_range(10000, 10005);
+			else
+				break;
+
+			if (i > 200)
 				break;
+			i++;
 		}
+
+		val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & 0x0003;
 		dev_dbg(codec->dev, "%s val = %d\n", __func__, val);
+
 		switch (val) {
 		case 1:
 		case 2:
-- 
cgit v1.2.3


From 1325734c97c533c2042794f2bd873486b4905cdb Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Fri, 7 Jul 2017 16:58:59 +0800
Subject: ASoC: rt5663: Modify the power sequence for reducing the pop sound

The patch modifies the power sequence for reducing the pop sound.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 36 +++++++++++++++++++++++++++++-------
 1 file changed, 29 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index ebd9ae00e04d..82edbcf39caa 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -2029,10 +2029,6 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 				RT5663_HP_SIG_SRC1_SILENCE);
 		} else {
 			snd_soc_write(codec, RT5663_DEPOP_2, 0x3003);
-			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x000b,
-				0x000b);
-			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030,
-				0x0030);
 			snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1,
 				RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_DIS);
 			snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_2, 0x1371);
@@ -2055,9 +2051,6 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0x0);
 			snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1,
 				RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN);
-			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0x0);
-			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x000b,
-				0x000b);
 		}
 		break;
 
@@ -2068,6 +2061,31 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rt5663_charge_pump_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (rt5663->codec_ver == CODEC_VER_0)
+			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x003b,
+				0x003b);
+		break;
+
+	case SND_SOC_DAPM_POST_PMD:
+		if (rt5663->codec_ver == CODEC_VER_0)
+			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x003b, 0);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static int rt5663_bst2_power(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -2187,6 +2205,9 @@ static const struct snd_soc_dapm_widget rt5663_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("DAC R", NULL, SND_SOC_NOPM, 0, 0),
 
 	/* Headphone*/
+	SND_SOC_DAPM_SUPPLY("HP Charge Pump", SND_SOC_NOPM, 0, 0,
+		rt5663_charge_pump_event, SND_SOC_DAPM_PRE_PMU |
+		SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5663_hp_event,
 		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 
@@ -2343,6 +2364,7 @@ static const struct snd_soc_dapm_route rt5663_dapm_routes[] = {
 	{ "STO1 DAC MIXR", NULL, "STO1 DAC R Power" },
 	{ "STO1 DAC MIXR", NULL, "STO1 DAC Filter" },
 
+	{ "HP Amp", NULL, "HP Charge Pump" },
 	{ "HP Amp", NULL, "DAC L" },
 	{ "HP Amp", NULL, "DAC R" },
 };
-- 
cgit v1.2.3


From 09b8852c759917e3ea10062b374692a5313cbfd8 Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Mon, 10 Jul 2017 16:20:45 +0100
Subject: ASoC: rt5663: add in missing loop counter to avoid infinite loop

The loop counter 'count' is missing an increment, so we could end up
with an infinite loop.  Add in a pre-increment to count to fix this.

Detected by CoverityScan, CID#1450120 ("Logically Dead Code")

Fixes: 1a425dbf1a10 ("ASoC: rt5663: Update the calibration funciton")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 82edbcf39caa..5b3c50c641d8 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -3044,7 +3044,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663)
 		else
 			break;
 
-		if (count > 200)
+		if (++count > 200)
 			break;
 	}
 
-- 
cgit v1.2.3


From 450f0f6a8fb499a6ef3d10ab5c29a7c4650a9958 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Mon, 10 Jul 2017 11:14:56 +0800
Subject: ASoC: rt5663: Add the manual offset field to compensate the DC offset

The patch adds the manual offset field in the devicetree to compensate the
DC offset that will be different between the PCB layout. It only can be
measured by the real production.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/rt5663.txt |  5 ++
 include/sound/rt5663.h                             | 20 ++++++++
 sound/soc/codecs/rt5663.c                          | 55 ++++++++++++++++++++--
 sound/soc/codecs/rt5663.h                          |  2 +
 4 files changed, 77 insertions(+), 5 deletions(-)
 create mode 100644 include/sound/rt5663.h

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt
index 70eaeaed2b18..f7c7cc513ee8 100644
--- a/Documentation/devicetree/bindings/sound/rt5663.txt
+++ b/Documentation/devicetree/bindings/sound/rt5663.txt
@@ -12,6 +12,11 @@ Required properties:
 
 Optional properties:
 
+- "realtek,dc_offset_l_manual"
+- "realtek,dc_offset_r_manual"
+  Based on the different PCB layout, add the manual offset value to
+  compensate the DC offset for each L and R channel.
+
 Pins on the device (for linking into audio routes) for RT5663:
 
   * IN1P
diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h
new file mode 100644
index 000000000000..be6f1b2d8fbd
--- /dev/null
+++ b/include/sound/rt5663.h
@@ -0,0 +1,20 @@
+/*
+ * linux/sound/rt5663.h -- Platform data for RT5663
+ *
+ * Copyright 2017 Realtek Semiconductor Corp.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5663_H
+#define __LINUX_SND_RT5663_H
+
+struct rt5663_platform_data {
+	unsigned int dc_offset_l_manual;
+	unsigned int dc_offset_r_manual;
+};
+
+#endif
+
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 5b3c50c641d8..963cda9c995a 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -40,6 +40,7 @@ enum {
 
 struct rt5663_priv {
 	struct snd_soc_codec *codec;
+	struct rt5663_platform_data pdata;
 	struct regmap *regmap;
 	struct delayed_work jack_detect_work;
 	struct snd_soc_jack *hs_jack;
@@ -57,6 +58,10 @@ struct rt5663_priv {
 	int jack_type;
 };
 
+static const struct reg_sequence rt5663_patch_list[] = {
+	{ 0x002a, 0x8080 },
+};
+
 static const struct reg_default rt5663_v2_reg[] = {
 	{ 0x0000, 0x0000 },
 	{ 0x0001, 0xc8c8 },
@@ -476,7 +481,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0023, 0x0039 },
 	{ 0x0026, 0xc0c0 },
 	{ 0x0029, 0x8080 },
-	{ 0x002a, 0xa0a0 },
+	{ 0x002a, 0x8080 },
 	{ 0x002c, 0x000c },
 	{ 0x002d, 0x0000 },
 	{ 0x0040, 0x0808 },
@@ -1958,15 +1963,11 @@ static const struct snd_kcontrol_new rt5663_adda_r_mix[] = {
 static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = {
 	SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER,
 			RT5663_M_DAC_L1_STO_L_SHIFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER,
-			RT5663_M_DAC_R1_STO_L_SHIFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = {
 	SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER,
 			RT5663_M_DAC_L1_STO_R_SHIFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER,
-			RT5663_M_DAC_R1_STO_R_SHIFT, 1, 1),
 };
 
 /* Out Switch */
@@ -3126,9 +3127,20 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663)
 	usleep_range(10000, 10005);
 }
 
+static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev)
+{
+	device_property_read_u32(dev, "realtek,dc_offset_l_manual",
+		&rt5663->pdata.dc_offset_l_manual);
+	device_property_read_u32(dev, "realtek,dc_offset_r_manual",
+		&rt5663->pdata.dc_offset_r_manual);
+
+	return 0;
+}
+
 static int rt5663_i2c_probe(struct i2c_client *i2c,
 		    const struct i2c_device_id *id)
 {
+	struct rt5663_platform_data *pdata = dev_get_platdata(&i2c->dev);
 	struct rt5663_priv *rt5663;
 	int ret;
 	unsigned int val;
@@ -3142,6 +3154,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, rt5663);
 
+	if (pdata)
+		rt5663->pdata = *pdata;
+	else
+		rt5663_parse_dp(rt5663, &i2c->dev);
+
 	regmap = devm_regmap_init_i2c(i2c, &temp_regmap);
 	if (IS_ERR(regmap)) {
 		ret = PTR_ERR(regmap);
@@ -3190,6 +3207,34 @@ static int rt5663_i2c_probe(struct i2c_client *i2c,
 	regmap_write(rt5663->regmap, RT5663_RESET, 0);
 	dev_dbg(&i2c->dev, "calibrate done\n");
 
+	switch (rt5663->codec_ver) {
+	case CODEC_VER_1:
+		break;
+	case CODEC_VER_0:
+		ret = regmap_register_patch(rt5663->regmap, rt5663_patch_list,
+					    ARRAY_SIZE(rt5663_patch_list));
+		if (ret != 0)
+			dev_warn(&i2c->dev,
+				"Failed to apply regmap patch: %d\n", ret);
+		break;
+	default:
+		dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__);
+	}
+
+	if (rt5663->pdata.dc_offset_l_manual) {
+		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2,
+			rt5663->pdata.dc_offset_l_manual >> 16);
+		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3,
+			rt5663->pdata.dc_offset_l_manual & 0xffff);
+	}
+
+	if (rt5663->pdata.dc_offset_r_manual) {
+		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5,
+			rt5663->pdata.dc_offset_r_manual >> 16);
+		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6,
+			rt5663->pdata.dc_offset_r_manual & 0xffff);
+	}
+
 	/* GPIO1 as IRQ */
 	regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5663_GP1_PIN_MASK,
 		RT5663_GP1_PIN_IRQ);
diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h
index 4621812c94d8..c5a9b69579ad 100644
--- a/sound/soc/codecs/rt5663.h
+++ b/sound/soc/codecs/rt5663.h
@@ -12,6 +12,8 @@
 #ifndef __RT5663_H__
 #define __RT5663_H__
 
+#include <sound/rt5663.h>
+
 /* Info */
 #define RT5663_RESET				0x0000
 #define RT5663_VENDOR_ID			0x00fd
-- 
cgit v1.2.3


From 4720c2fe196fc16fca9c902bec5c786ae4689842 Mon Sep 17 00:00:00 2001
From: Jaechul Lee
Date: Fri, 7 Jul 2017 10:31:10 +0900
Subject: ASoC: samsung: i2s: Support more resolution rates

This driver can support more frequencies over 96KHz. There are no reasons
to limit the frequency range below 96KHz. If codecs/amps or something else
can't support higher resolution rates, the constraints would be set rates
properly because each drivers have its own limits.

I added the 'pcm_rates' field to the dai_data to be set rates by the
compatibilities. As a result, rates will be set each devices respectively.
For example of exynos5433, rates will be set from 8KHz to 192KHz.

Signed-off-by: Jaechul Lee <jcsing.lee@samsung.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/i2s.c | 20 +++++++++++++-------
 1 file changed, 13 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index af3ba4d4ccc5..c9f87f7bae99 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -50,6 +50,7 @@ struct samsung_i2s_variant_regs {
 
 struct samsung_i2s_dai_data {
 	u32 quirks;
+	unsigned int pcm_rates;
 	const struct samsung_i2s_variant_regs *i2s_variant_regs;
 };
 
@@ -1076,13 +1077,13 @@ static const struct snd_soc_component_driver samsung_i2s_component = {
 	.name		= "samsung-i2s",
 };
 
-#define SAMSUNG_I2S_RATES	SNDRV_PCM_RATE_8000_96000
-
 #define SAMSUNG_I2S_FMTS	(SNDRV_PCM_FMTBIT_S8 | \
 					SNDRV_PCM_FMTBIT_S16_LE | \
 					SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
+static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev,
+				const struct samsung_i2s_dai_data *i2s_dai_data,
+				bool sec)
 {
 	struct i2s_dai *i2s;
 
@@ -1101,13 +1102,13 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
 	i2s->i2s_dai_drv.resume = i2s_resume;
 	i2s->i2s_dai_drv.playback.channels_min = 1;
 	i2s->i2s_dai_drv.playback.channels_max = 2;
-	i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES;
+	i2s->i2s_dai_drv.playback.rates = i2s_dai_data->pcm_rates;
 	i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS;
 
 	if (!sec) {
 		i2s->i2s_dai_drv.capture.channels_min = 1;
 		i2s->i2s_dai_drv.capture.channels_max = 2;
-		i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES;
+		i2s->i2s_dai_drv.capture.rates = i2s_dai_data->pcm_rates;
 		i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS;
 	}
 	return i2s;
@@ -1242,7 +1243,7 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 		i2s_dai_data = (struct samsung_i2s_dai_data *)
 				platform_get_device_id(pdev)->driver_data;
 
-	pri_dai = i2s_alloc_dai(pdev, false);
+	pri_dai = i2s_alloc_dai(pdev, i2s_dai_data, false);
 	if (!pri_dai) {
 		dev_err(&pdev->dev, "Unable to alloc I2S_pri\n");
 		return -ENOMEM;
@@ -1316,7 +1317,7 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 		goto err_disable_clk;
 
 	if (quirks & QUIRK_SEC_DAI) {
-		sec_dai = i2s_alloc_dai(pdev, true);
+		sec_dai = i2s_alloc_dai(pdev, i2s_dai_data, true);
 		if (!sec_dai) {
 			dev_err(&pdev->dev, "Unable to alloc I2S_sec\n");
 			ret = -ENOMEM;
@@ -1452,29 +1453,34 @@ static const struct samsung_i2s_variant_regs i2sv5_i2s1_regs = {
 
 static const struct samsung_i2s_dai_data i2sv3_dai_type = {
 	.quirks = QUIRK_NO_MUXPSR,
+	.pcm_rates = SNDRV_PCM_RATE_8000_96000,
 	.i2s_variant_regs = &i2sv3_regs,
 };
 
 static const struct samsung_i2s_dai_data i2sv5_dai_type = {
 	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
 			QUIRK_SUPPORTS_IDMA,
+	.pcm_rates = SNDRV_PCM_RATE_8000_96000,
 	.i2s_variant_regs = &i2sv3_regs,
 };
 
 static const struct samsung_i2s_dai_data i2sv6_dai_type = {
 	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
 			QUIRK_SUPPORTS_TDM | QUIRK_SUPPORTS_IDMA,
+	.pcm_rates = SNDRV_PCM_RATE_8000_96000,
 	.i2s_variant_regs = &i2sv6_regs,
 };
 
 static const struct samsung_i2s_dai_data i2sv7_dai_type = {
 	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
 			QUIRK_SUPPORTS_TDM,
+	.pcm_rates = SNDRV_PCM_RATE_8000_192000,
 	.i2s_variant_regs = &i2sv7_regs,
 };
 
 static const struct samsung_i2s_dai_data i2sv5_dai_type_i2s1 = {
 	.quirks = QUIRK_PRI_6CHAN | QUIRK_NEED_RSTCLR,
+	.pcm_rates = SNDRV_PCM_RATE_8000_96000,
 	.i2s_variant_regs = &i2sv5_i2s1_regs,
 };
 
-- 
cgit v1.2.3


From 27d30400c448264c1ac9434cb836de0c230af213 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Fri, 30 Jun 2017 15:43:50 -0500
Subject: ASoC: spear: fix error return code in spdif_in_probe()

platform_get_irq() returns an error code, but the spdif_in driver
ignores it and always returns -EINVAL. This is not correct, and
prevents -EPROBE_DEFER from being propagated properly.

Notice that platform_get_irq() no longer returns 0 on error:
https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?id=e330b9a6bb35dc7097a4f02cb1ae7b6f96df92af

Print error message and propagate the return value of platform_get_irq
on failure.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/spear/spdif_in.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 977a078eb92f..7f32527fc3c8 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -223,8 +223,10 @@ static int spdif_in_probe(struct platform_device *pdev)
 
 	host->io_base = io_base;
 	host->irq = platform_get_irq(pdev, 0);
-	if (host->irq < 0)
-		return -EINVAL;
+	if (host->irq < 0) {
+		dev_warn(&pdev->dev, "failed to get IRQ: %d\n", host->irq);
+		return host->irq;
+	}
 
 	host->clk = devm_clk_get(&pdev->dev, NULL);
 	if (IS_ERR(host->clk))
-- 
cgit v1.2.3


From cfe267da9de756d15eab095358b0b122c57816a6 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Fri, 30 Jun 2017 17:26:48 -0500
Subject: ASoC: twl6040: fix error return code in twl6040_probe()

platform_get_irq() returns an error code, but the twl6040
driver ignores it and always returns -EINVAL. This is not correct,
and prevents -EPROBE_DEFER from being propagated properly.
Notice that platform_get_irq() no longer returns 0 on error.

Print and propagate the return value of platform_get_irq on failure.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/twl6040.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 2b6ad09e0886..96cf94a2b7a0 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1123,8 +1123,8 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 
 	priv->plug_irq = platform_get_irq(pdev, 0);
 	if (priv->plug_irq < 0) {
-		dev_err(codec->dev, "invalid irq\n");
-		return -EINVAL;
+		dev_err(codec->dev, "invalid irq: %d\n", priv->plug_irq);
+		return priv->plug_irq;
 	}
 
 	INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work);
-- 
cgit v1.2.3


From 2d397c518eead8b6c3cd817c29bc2d4fb1c41830 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski
Date: Mon, 17 Jul 2017 07:44:21 +0200
Subject: ASoC: samsung: Remove non-existing CONFIG_CPU_S3C2413

There is no CONFIG_CPU_S3C2413 so get rid of it.

Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/s3c-i2s-v2.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 8f42deaa184b..ca522a95160b 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -27,7 +27,7 @@
 
 #undef S3C_IIS_V2_SUPPORTED
 
-#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \
+#if defined(CONFIG_CPU_S3C2412) \
 	|| defined(CONFIG_ARCH_S3C64XX) || defined(CONFIG_CPU_S5PV210)
 #define S3C_IIS_V2_SUPPORTED
 #endif
-- 
cgit v1.2.3


From 4b2983da22d30c53d778b2e5035dc347dc4b0037 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Sun, 16 Jul 2017 21:12:21 -0500
Subject: ASoC: fsi: constify dev_pm_ops structure

This dev_pm_ops structure is only stored in the pm field of a
device_driver structure. This field is declared const, so
dev_pm_ops structures that have this property can be declared
as const also.

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/fsi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 7c4bdd82bb95..005b21506556 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -2109,7 +2109,7 @@ static int fsi_resume(struct device *dev)
 	return 0;
 }
 
-static struct dev_pm_ops fsi_pm_ops = {
+static const struct dev_pm_ops fsi_pm_ops = {
 	.suspend		= fsi_suspend,
 	.resume			= fsi_resume,
 };
-- 
cgit v1.2.3


From da61a11823395c87db0471d8d54055fa5a961688 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Sun, 16 Jul 2017 21:12:39 -0500
Subject: ASoC: psc-ac97: constify dev_pm_ops structure

This dev_pm_ops structure is only stored in the pm field of a
device_driver structure. This field is declared const, so
dev_pm_ops structures that have this property can be declared
as const also.

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/au1x/psc-ac97.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index bb53c7059005..a2050ae5a3fe 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -474,7 +474,7 @@ static int au1xpsc_ac97_drvresume(struct device *dev)
 	return 0;
 }
 
-static struct dev_pm_ops au1xpscac97_pmops = {
+static const struct dev_pm_ops au1xpscac97_pmops = {
 	.suspend	= au1xpsc_ac97_drvsuspend,
 	.resume		= au1xpsc_ac97_drvresume,
 };
-- 
cgit v1.2.3


From 90982dcdda4f0163bc5e53f49301e155c2cd8547 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Sun, 16 Jul 2017 21:12:49 -0500
Subject: ASoC: au1x: psc-i2s: constify dev_pm_ops structure

This dev_pm_ops structure is only stored in the pm field of a
device_driver structure. This field is declared const, so
dev_pm_ops structures that have this property can be declared
as const also.

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/au1x/psc-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 0bf9d62b91a0..e6eec081eaae 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -392,7 +392,7 @@ static int au1xpsc_i2s_drvresume(struct device *dev)
 	return 0;
 }
 
-static struct dev_pm_ops au1xpsci2s_pmops = {
+static const struct dev_pm_ops au1xpsci2s_pmops = {
 	.suspend	= au1xpsc_i2s_drvsuspend,
 	.resume		= au1xpsc_i2s_drvresume,
 };
-- 
cgit v1.2.3


From d5c6d43b3d94e7bf421ec2aaf21b4b7c79fa1850 Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Fri, 14 Jul 2017 00:04:06 +0100
Subject: ASoC: Intel: Skylake: fix type in debug message

Trivial fix to typo in debug message, dst_instacne should be
dst_instance

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index fb2f1f603f3c..c5b4e0a9946c 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -950,7 +950,7 @@ static void skl_dump_bind_info(struct skl_sst *ctx, struct skl_module_cfg
 {
 	dev_dbg(ctx->dev, "%s: src module_id = %d  src_instance=%d\n",
 		__func__, src_module->id.module_id, src_module->id.pvt_id);
-	dev_dbg(ctx->dev, "%s: dst_module=%d dst_instacne=%d\n", __func__,
+	dev_dbg(ctx->dev, "%s: dst_module=%d dst_instance=%d\n", __func__,
 		 dst_module->id.module_id, dst_module->id.pvt_id);
 
 	dev_dbg(ctx->dev, "src_module state = %d dst module state = %d\n",
-- 
cgit v1.2.3


From 82e2b1e081906d3d86c4597250ab2939b209079f Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 01:16:46 -0500
Subject: ASoC: Intel: Skylake: constify snd_soc_dai_ops structures

These structures are only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-pcm.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 0ebea34a4988..b1f2dba642c4 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -628,7 +628,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops skl_pcm_dai_ops = {
+static const struct snd_soc_dai_ops skl_pcm_dai_ops = {
 	.startup = skl_pcm_open,
 	.shutdown = skl_pcm_close,
 	.prepare = skl_pcm_prepare,
@@ -637,15 +637,15 @@ static struct snd_soc_dai_ops skl_pcm_dai_ops = {
 	.trigger = skl_pcm_trigger,
 };
 
-static struct snd_soc_dai_ops skl_dmic_dai_ops = {
+static const struct snd_soc_dai_ops skl_dmic_dai_ops = {
 	.hw_params = skl_be_hw_params,
 };
 
-static struct snd_soc_dai_ops skl_be_ssp_dai_ops = {
+static const struct snd_soc_dai_ops skl_be_ssp_dai_ops = {
 	.hw_params = skl_be_hw_params,
 };
 
-static struct snd_soc_dai_ops skl_link_dai_ops = {
+static const struct snd_soc_dai_ops skl_link_dai_ops = {
 	.prepare = skl_link_pcm_prepare,
 	.hw_params = skl_link_hw_params,
 	.hw_free = skl_link_hw_free,
-- 
cgit v1.2.3


From c115a3178b1dfacd57b643934bb6bbbd1a9557f0 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 01:26:06 -0500
Subject: ASoC: Intel: Atom: constify snd_soc_dai_ops structures

These structures are only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index b082b31023d5..49c7b88b5bdc 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -471,7 +471,7 @@ static void sst_disable_ssp(struct snd_pcm_substream *substream,
 	}
 }
 
-static struct snd_soc_dai_ops sst_media_dai_ops = {
+static const struct snd_soc_dai_ops sst_media_dai_ops = {
 	.startup = sst_media_open,
 	.shutdown = sst_media_close,
 	.prepare = sst_media_prepare,
@@ -480,11 +480,11 @@ static struct snd_soc_dai_ops sst_media_dai_ops = {
 	.mute_stream = sst_media_digital_mute,
 };
 
-static struct snd_soc_dai_ops sst_compr_dai_ops = {
+static const struct snd_soc_dai_ops sst_compr_dai_ops = {
 	.mute_stream = sst_media_digital_mute,
 };
 
-static struct snd_soc_dai_ops sst_be_dai_ops = {
+static const struct snd_soc_dai_ops sst_be_dai_ops = {
 	.startup = sst_enable_ssp,
 	.hw_params = sst_be_hw_params,
 	.set_fmt = sst_set_format,
-- 
cgit v1.2.3


From fa05899c12d54d3a1befa742f62d3188e551a87c Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Fri, 14 Jul 2017 14:36:32 +0800
Subject: ASoC: rt5665: force using PLL if MCLK is not suitable

Idealy PLL will be set in machine driver if MCLK doesn't meet the
requirement of codec. To let the codec driver be more general, we
can use a common PLL setting once sysclk/pll doesn't set properly
in machine driver.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5665.c | 9 ++++++---
 1 file changed, 6 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index 370ed54d1e15..d3103efcb135 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -4098,9 +4098,12 @@ static int rt5665_hw_params(struct snd_pcm_substream *substream,
 	rt5665->lrck[dai->id] = params_rate(params);
 	pre_div = rl6231_get_clk_info(rt5665->sysclk, rt5665->lrck[dai->id]);
 	if (pre_div < 0) {
-		dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n",
-			rt5665->lrck[dai->id], dai->id);
-		return -EINVAL;
+		dev_warn(codec->dev, "Force using PLL");
+		snd_soc_codec_set_pll(codec, 0, RT5665_PLL1_S_MCLK,
+			rt5665->sysclk,	rt5665->lrck[dai->id] * 512);
+		snd_soc_codec_set_sysclk(codec, RT5665_SCLK_S_PLL1, 0,
+			rt5665->lrck[dai->id] * 512, 0);
+		pre_div = 1;
 	}
 	frame_size = snd_soc_params_to_frame_size(params);
 	if (frame_size < 0) {
-- 
cgit v1.2.3


From 53ba2aa314164f7fcd3acf9594313811256a52c4 Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Sat, 15 Jul 2017 09:19:07 +0200
Subject: ASoC: rsnd: add missing of_node_put

for_each_child_of_node performs an of_node_get on each iteration, so a
jump out of the loop requires an of_node_put.

The semantic patch that fixes this problem is as follows
(http://coccinelle.lip6.fr):

// <smpl>
@@
local idexpression n;
expression e,e1;
identifier l;
@@

 for_each_child_of_node(e1,n) {
   ...
(
   of_node_put(n);
|
   e = n
|
+  of_node_put(n);
?  goto l;
)
   ...
 }
...
l: ... when != n
// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ctu.c | 5 ++++-
 sound/soc/sh/rcar/dvc.c | 5 ++++-
 sound/soc/sh/rcar/mix.c | 5 ++++-
 sound/soc/sh/rcar/src.c | 6 +++++-
 sound/soc/sh/rcar/ssi.c | 6 +++++-
 5 files changed, 22 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 4ba8f2fe7a4c..e7f53f44165d 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -394,13 +394,16 @@ int rsnd_ctu_probe(struct rsnd_priv *priv)
 		clk = devm_clk_get(dev, name);
 		if (IS_ERR(clk)) {
 			ret = PTR_ERR(clk);
+			of_node_put(np);
 			goto rsnd_ctu_probe_done;
 		}
 
 		ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops,
 				    clk, rsnd_mod_get_status, RSND_MOD_CTU, i);
-		if (ret)
+		if (ret) {
+			of_node_put(np);
 			goto rsnd_ctu_probe_done;
+		}
 
 		i++;
 	}
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index 99d2d9459e75..1743ade3cc55 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -380,13 +380,16 @@ int rsnd_dvc_probe(struct rsnd_priv *priv)
 		clk = devm_clk_get(dev, name);
 		if (IS_ERR(clk)) {
 			ret = PTR_ERR(clk);
+			of_node_put(np);
 			goto rsnd_dvc_probe_done;
 		}
 
 		ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops,
 				    clk, rsnd_mod_get_status, RSND_MOD_DVC, i);
-		if (ret)
+		if (ret) {
+			of_node_put(np);
 			goto rsnd_dvc_probe_done;
+		}
 
 		i++;
 	}
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 195fc7bb22af..6c4826c189a4 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -168,13 +168,16 @@ int rsnd_mix_probe(struct rsnd_priv *priv)
 		clk = devm_clk_get(dev, name);
 		if (IS_ERR(clk)) {
 			ret = PTR_ERR(clk);
+			of_node_put(np);
 			goto rsnd_mix_probe_done;
 		}
 
 		ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops,
 				    clk, rsnd_mod_get_status, RSND_MOD_MIX, i);
-		if (ret)
+		if (ret) {
+			of_node_put(np);
 			goto rsnd_mix_probe_done;
+		}
 
 		i++;
 	}
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 7aa239e28491..8e9a1de0ec25 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -581,20 +581,24 @@ int rsnd_src_probe(struct rsnd_priv *priv)
 		src->irq = irq_of_parse_and_map(np, 0);
 		if (!src->irq) {
 			ret = -EINVAL;
+			of_node_put(np);
 			goto rsnd_src_probe_done;
 		}
 
 		clk = devm_clk_get(dev, name);
 		if (IS_ERR(clk)) {
 			ret = PTR_ERR(clk);
+			of_node_put(np);
 			goto rsnd_src_probe_done;
 		}
 
 		ret = rsnd_mod_init(priv, rsnd_mod_get(src),
 				    &rsnd_src_ops, clk, rsnd_mod_get_status,
 				    RSND_MOD_SRC, i);
-		if (ret)
+		if (ret) {
+			of_node_put(np);
 			goto rsnd_src_probe_done;
+		}
 
 skip:
 		i++;
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 46feddd78ee2..49cdc5e007b3 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -1079,6 +1079,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv)
 		clk = devm_clk_get(dev, name);
 		if (IS_ERR(clk)) {
 			ret = PTR_ERR(clk);
+			of_node_put(np);
 			goto rsnd_ssi_probe_done;
 		}
 
@@ -1091,6 +1092,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv)
 		ssi->irq = irq_of_parse_and_map(np, 0);
 		if (!ssi->irq) {
 			ret = -EINVAL;
+			of_node_put(np);
 			goto rsnd_ssi_probe_done;
 		}
 
@@ -1101,8 +1103,10 @@ int rsnd_ssi_probe(struct rsnd_priv *priv)
 
 		ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk,
 				    rsnd_ssi_get_status, RSND_MOD_SSI, i);
-		if (ret)
+		if (ret) {
+			of_node_put(np);
 			goto rsnd_ssi_probe_done;
+		}
 
 		i++;
 	}
-- 
cgit v1.2.3


From d51579e6effa19c1ca4d6a29c3841cb454d56af6 Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:06 -0300
Subject: ASoC: sun4i-codec: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-codec.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 150069987c0c..4c37231f254b 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -1171,9 +1171,8 @@ static int sun4i_codec_spk_event(struct snd_soc_dapm_widget *w,
 {
 	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(w->dapm->card);
 
-	if (scodec->gpio_pa)
-		gpiod_set_value_cansleep(scodec->gpio_pa,
-					 !!SND_SOC_DAPM_EVENT_ON(event));
+	gpiod_set_value_cansleep(scodec->gpio_pa,
+				 !!SND_SOC_DAPM_EVENT_ON(event));
 
 	return 0;
 }
-- 
cgit v1.2.3


From 68feed90070e5133c0bfe26648410a39e092018d Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:08 -0300
Subject: ASoC: adau1977: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/adau1977.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index b319db6a69f8..ccc5cf0e4f80 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -388,8 +388,7 @@ static int adau1977_power_disable(struct adau1977 *adau1977)
 
 	regcache_mark_dirty(adau1977->regmap);
 
-	if (adau1977->reset_gpio)
-		gpiod_set_value_cansleep(adau1977->reset_gpio, 0);
+	gpiod_set_value_cansleep(adau1977->reset_gpio, 0);
 
 	regcache_cache_only(adau1977->regmap, true);
 
@@ -420,8 +419,7 @@ static int adau1977_power_enable(struct adau1977 *adau1977)
 			goto err_disable_avdd;
 	}
 
-	if (adau1977->reset_gpio)
-		gpiod_set_value_cansleep(adau1977->reset_gpio, 1);
+	gpiod_set_value_cansleep(adau1977->reset_gpio, 1);
 
 	regcache_cache_only(adau1977->regmap, false);
 
-- 
cgit v1.2.3


From 89b13706c4bb6dae0263014051dd59986376faa5 Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:07 -0300
Subject: ASoC: wm8804: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8804.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index af95d648265b..fc69b87443d8 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -623,8 +623,7 @@ int wm8804_probe(struct device *dev, struct regmap *regmap)
 		return ret;
 	}
 
-	if (wm8804->reset)
-		gpiod_set_value_cansleep(wm8804->reset, 1);
+	gpiod_set_value_cansleep(wm8804->reset, 1);
 
 	ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1);
 	if (ret < 0) {
-- 
cgit v1.2.3


From c5660778b4851b1aea5d848ccfde3f27949f6418 Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:09 -0300
Subject: ASoC: cs35l33: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs35l33.c | 12 ++++--------
 1 file changed, 4 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c
index 6df29fa30fb9..1eea91271ebb 100644
--- a/sound/soc/codecs/cs35l33.c
+++ b/sound/soc/codecs/cs35l33.c
@@ -869,8 +869,7 @@ static int __maybe_unused cs35l33_runtime_resume(struct device *dev)
 
 	dev_dbg(dev, "%s\n", __func__);
 
-	if (cs35l33->reset_gpio)
-		gpiod_set_value_cansleep(cs35l33->reset_gpio, 0);
+	gpiod_set_value_cansleep(cs35l33->reset_gpio, 0);
 
 	ret = regulator_bulk_enable(cs35l33->num_core_supplies,
 		cs35l33->core_supplies);
@@ -881,8 +880,7 @@ static int __maybe_unused cs35l33_runtime_resume(struct device *dev)
 
 	regcache_cache_only(cs35l33->regmap, false);
 
-	if (cs35l33->reset_gpio)
-		gpiod_set_value_cansleep(cs35l33->reset_gpio, 1);
+	gpiod_set_value_cansleep(cs35l33->reset_gpio, 1);
 
 	msleep(CS35L33_BOOT_DELAY);
 
@@ -1191,8 +1189,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client,
 		return ret;
 	}
 
-	if (cs35l33->reset_gpio)
-		gpiod_set_value_cansleep(cs35l33->reset_gpio, 1);
+	gpiod_set_value_cansleep(cs35l33->reset_gpio, 1);
 
 	msleep(CS35L33_BOOT_DELAY);
 	regcache_cache_only(cs35l33->regmap, false);
@@ -1262,8 +1259,7 @@ static int cs35l33_i2c_remove(struct i2c_client *client)
 
 	snd_soc_unregister_codec(&client->dev);
 
-	if (cs35l33->reset_gpio)
-		gpiod_set_value_cansleep(cs35l33->reset_gpio, 0);
+	gpiod_set_value_cansleep(cs35l33->reset_gpio, 0);
 
 	pm_runtime_disable(&client->dev);
 	regulator_bulk_disable(cs35l33->num_core_supplies,
-- 
cgit v1.2.3


From db3be255a101da11f67c0f3053b750e767eb10fe Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:10 -0300
Subject: ASoC: cs35l34: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs35l34.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c
index 0a747c66cc6c..dade899b38dc 100644
--- a/sound/soc/codecs/cs35l34.c
+++ b/sound/soc/codecs/cs35l34.c
@@ -1138,8 +1138,7 @@ static int cs35l34_i2c_remove(struct i2c_client *client)
 
 	snd_soc_unregister_codec(&client->dev);
 
-	if (cs35l34->reset_gpio)
-		gpiod_set_value_cansleep(cs35l34->reset_gpio, 0);
+	gpiod_set_value_cansleep(cs35l34->reset_gpio, 0);
 
 	pm_runtime_disable(&client->dev);
 	regulator_bulk_disable(cs35l34->num_core_supplies,
-- 
cgit v1.2.3


From 205cbac065f9589c77a79f26339cf63023f08f6e Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:11 -0300
Subject: ASoC: cs42l42: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 55e4520cdcaf..55caf5ff3e64 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -1898,8 +1898,7 @@ static int cs42l42_i2c_remove(struct i2c_client *i2c_client)
 	snd_soc_unregister_codec(&i2c_client->dev);
 
 	/* Hold down reset */
-	if (cs42l42->reset_gpio)
-		gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
+	gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
 
 	return 0;
 }
@@ -1913,8 +1912,7 @@ static int cs42l42_runtime_suspend(struct device *dev)
 	regcache_mark_dirty(cs42l42->regmap);
 
 	/* Hold down reset */
-	if (cs42l42->reset_gpio)
-		gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
+	gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
 
 	/* remove power */
 	regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies),
@@ -1937,8 +1935,7 @@ static int cs42l42_runtime_resume(struct device *dev)
 		return ret;
 	}
 
-	if (cs42l42->reset_gpio)
-		gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+	gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
 
 	regcache_cache_only(cs42l42->regmap, false);
 	regcache_sync(cs42l42->regmap);
-- 
cgit v1.2.3


From 55be663dc71ec50f0e383c50886cbd3be69fcd2e Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:12 -0300
Subject: ASoC: cs53l30: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs53l30.c | 15 +++++----------
 1 file changed, 5 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c
index 06933a5d0a75..92fbe6303dd4 100644
--- a/sound/soc/codecs/cs53l30.c
+++ b/sound/soc/codecs/cs53l30.c
@@ -842,8 +842,7 @@ static int cs53l30_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 {
 	struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec);
 
-	if (priv->mute_gpio)
-		gpiod_set_value_cansleep(priv->mute_gpio, mute);
+	gpiod_set_value_cansleep(priv->mute_gpio, mute);
 
 	return 0;
 }
@@ -960,8 +959,7 @@ static int cs53l30_i2c_probe(struct i2c_client *client,
 		goto error;
 	}
 
-	if (cs53l30->reset_gpio)
-		gpiod_set_value_cansleep(cs53l30->reset_gpio, 1);
+	gpiod_set_value_cansleep(cs53l30->reset_gpio, 1);
 
 	i2c_set_clientdata(client, cs53l30);
 
@@ -1056,8 +1054,7 @@ static int cs53l30_i2c_remove(struct i2c_client *client)
 	snd_soc_unregister_codec(&client->dev);
 
 	/* Hold down reset */
-	if (cs53l30->reset_gpio)
-		gpiod_set_value_cansleep(cs53l30->reset_gpio, 0);
+	gpiod_set_value_cansleep(cs53l30->reset_gpio, 0);
 
 	regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies),
 			       cs53l30->supplies);
@@ -1073,8 +1070,7 @@ static int cs53l30_runtime_suspend(struct device *dev)
 	regcache_cache_only(cs53l30->regmap, true);
 
 	/* Hold down reset */
-	if (cs53l30->reset_gpio)
-		gpiod_set_value_cansleep(cs53l30->reset_gpio, 0);
+	gpiod_set_value_cansleep(cs53l30->reset_gpio, 0);
 
 	regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies),
 			       cs53l30->supplies);
@@ -1094,8 +1090,7 @@ static int cs53l30_runtime_resume(struct device *dev)
 		return ret;
 	}
 
-	if (cs53l30->reset_gpio)
-		gpiod_set_value_cansleep(cs53l30->reset_gpio, 1);
+	gpiod_set_value_cansleep(cs53l30->reset_gpio, 1);
 
 	regcache_cache_only(cs53l30->regmap, false);
 	ret = regcache_sync(cs53l30->regmap);
-- 
cgit v1.2.3


From c5efe232e09ef29cc2f9e8333a49dd8891366c22 Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sun, 16 Jul 2017 18:11:13 -0300
Subject: ASoC: sta32x: Remove unneeded gpiod NULL check

The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sta32x.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 0790ae8530d9..5b888476d9ff 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -847,8 +847,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
 		msleep(300);
 		sta32x_watchdog_stop(sta32x);
 
-		if (sta32x->gpiod_nreset)
-			gpiod_set_value(sta32x->gpiod_nreset, 0);
+		gpiod_set_value(sta32x->gpiod_nreset, 0);
 
 		regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies),
 				       sta32x->supplies);
-- 
cgit v1.2.3


From 173f4612690ac569d9f912a31aa69af15b84047c Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Thu, 13 Jul 2017 19:42:20 +0800
Subject: ASoC: rt5514: Support the DSP recording continuously after the
 hotwording triggered

The patch uses the IRQ to copy the PCM data to userspace continuously after
the hotwording triggered from DSP.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514-spi.c | 139 ++++++++++++++++++++++--------------------
 sound/soc/codecs/rt5514-spi.h |   7 +--
 2 files changed, 76 insertions(+), 70 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 7ed62e8c80b4..950d1ffdc06c 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -43,9 +43,7 @@ struct rt5514_dsp {
 	struct mutex dma_lock;
 	struct snd_pcm_substream *substream;
 	unsigned int buf_base, buf_limit, buf_rp;
-	size_t buf_size;
-	size_t dma_offset;
-	size_t dsp_offset;
+	size_t buf_size, get_size, dma_offset;
 };
 
 static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = {
@@ -80,6 +78,8 @@ static void rt5514_spi_copy_work(struct work_struct *work)
 		container_of(work, struct rt5514_dsp, copy_work.work);
 	struct snd_pcm_runtime *runtime;
 	size_t period_bytes, truncated_bytes = 0;
+	unsigned int cur_wp, remain_data;
+	u8 buf[8];
 
 	mutex_lock(&rt5514_dsp->dma_lock);
 	if (!rt5514_dsp->substream) {
@@ -90,8 +90,24 @@ static void rt5514_spi_copy_work(struct work_struct *work)
 	runtime = rt5514_dsp->substream->runtime;
 	period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
 
-	if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset <  period_bytes)
-		period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset;
+	if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) {
+		rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf,
+			sizeof(buf));
+		cur_wp = buf[0] | buf[1] << 8 | buf[2] << 16 |
+					buf[3] << 24;
+
+		if (cur_wp >= rt5514_dsp->buf_rp)
+			remain_data = (cur_wp - rt5514_dsp->buf_rp);
+		else
+			remain_data =
+				(rt5514_dsp->buf_limit - rt5514_dsp->buf_rp) +
+				(cur_wp - rt5514_dsp->buf_base);
+
+		if (remain_data < period_bytes) {
+			schedule_delayed_work(&rt5514_dsp->copy_work, 5);
+			goto done;
+		}
+	}
 
 	if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) {
 		rt5514_spi_burst_read(rt5514_dsp->buf_rp,
@@ -112,24 +128,58 @@ static void rt5514_spi_copy_work(struct work_struct *work)
 			runtime->dma_area + rt5514_dsp->dma_offset +
 			truncated_bytes, period_bytes - truncated_bytes);
 
-			rt5514_dsp->buf_rp = rt5514_dsp->buf_base +
-				period_bytes - truncated_bytes;
+		rt5514_dsp->buf_rp = rt5514_dsp->buf_base + period_bytes -
+			truncated_bytes;
 	}
 
+	rt5514_dsp->get_size += period_bytes;
 	rt5514_dsp->dma_offset += period_bytes;
 	if (rt5514_dsp->dma_offset >= runtime->dma_bytes)
 		rt5514_dsp->dma_offset = 0;
 
-	rt5514_dsp->dsp_offset += period_bytes;
-
 	snd_pcm_period_elapsed(rt5514_dsp->substream);
 
-	if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size)
-		schedule_delayed_work(&rt5514_dsp->copy_work, 5);
+	schedule_delayed_work(&rt5514_dsp->copy_work, 5);
+
 done:
 	mutex_unlock(&rt5514_dsp->dma_lock);
 }
 
+static irqreturn_t rt5514_spi_irq(int irq, void *data)
+{
+	struct rt5514_dsp *rt5514_dsp = data;
+	u8 buf[8];
+
+	rt5514_dsp->get_size = 0;
+	rt5514_dsp->dma_offset = 0;
+
+	/**
+	 * The address area x1800XXXX is the register address, and it cannot
+	 * support spi burst read perfectly. So we use the spi burst read
+	 * individually to make sure the data correctly.
+	 */
+	rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf,
+		sizeof(buf));
+	rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 |
+				buf[3] << 24;
+
+	rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf,
+		sizeof(buf));
+	rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 |
+				buf[3] << 24;
+
+	rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf,
+		sizeof(buf));
+	rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 |
+				buf[3] << 24;
+
+	rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base;
+
+	schedule_delayed_work(&rt5514_dsp->copy_work, 0);
+
+	return IRQ_HANDLED;
+}
+
 /* PCM for streaming audio from the DSP buffer */
 static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream)
 {
@@ -170,59 +220,6 @@ static int rt5514_spi_hw_free(struct snd_pcm_substream *substream)
 	return snd_pcm_lib_free_vmalloc_buffer(substream);
 }
 
-static int rt5514_spi_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct rt5514_dsp *rt5514_dsp =
-			snd_soc_platform_get_drvdata(rtd->platform);
-	u8 buf[8];
-
-	rt5514_dsp->dma_offset = 0;
-	rt5514_dsp->dsp_offset = 0;
-
-	/**
-	 * The address area x1800XXXX is the register address, and it cannot
-	 * support spi burst read perfectly. So we use the spi burst read
-	 * individually to make sure the data correctly.
-	*/
-	rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf,
-		sizeof(buf));
-	rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 |
-				buf[3] << 24;
-
-	rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf,
-		sizeof(buf));
-	rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 |
-				buf[3] << 24;
-
-	rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf,
-		sizeof(buf));
-	rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 |
-				buf[3] << 24;
-
-	rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf,
-		sizeof(buf));
-	rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 |
-				buf[3] << 24;
-
-	return 0;
-}
-
-static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct rt5514_dsp *rt5514_dsp =
-			snd_soc_platform_get_drvdata(rtd->platform);
-
-	if (cmd == SNDRV_PCM_TRIGGER_START) {
-		if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit &&
-			rt5514_dsp->buf_rp && rt5514_dsp->buf_size)
-			schedule_delayed_work(&rt5514_dsp->copy_work, 0);
-	}
-
-	return 0;
-}
-
 static snd_pcm_uframes_t rt5514_spi_pcm_pointer(
 		struct snd_pcm_substream *substream)
 {
@@ -238,8 +235,6 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = {
 	.open		= rt5514_spi_pcm_open,
 	.hw_params	= rt5514_spi_hw_params,
 	.hw_free	= rt5514_spi_hw_free,
-	.trigger	= rt5514_spi_trigger,
-	.prepare	= rt5514_spi_prepare,
 	.pointer	= rt5514_spi_pcm_pointer,
 	.mmap		= snd_pcm_lib_mmap_vmalloc,
 	.page		= snd_pcm_lib_get_vmalloc_page,
@@ -248,6 +243,7 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = {
 static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform)
 {
 	struct rt5514_dsp *rt5514_dsp;
+	int ret;
 
 	rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp),
 			GFP_KERNEL);
@@ -257,6 +253,17 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform)
 	INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work);
 	snd_soc_platform_set_drvdata(platform, rt5514_dsp);
 
+	if (rt5514_spi->irq) {
+		ret = devm_request_threaded_irq(&rt5514_spi->dev,
+			rt5514_spi->irq, NULL, rt5514_spi_irq,
+			IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5514-spi",
+			rt5514_dsp);
+		if (ret)
+			dev_err(&rt5514_spi->dev,
+				"%s Failed to reguest IRQ: %d\n", __func__,
+				ret);
+	}
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h
index f69b1cdf2f9b..a6434ee6ff03 100644
--- a/sound/soc/codecs/rt5514-spi.h
+++ b/sound/soc/codecs/rt5514-spi.h
@@ -17,10 +17,9 @@
 */
 #define RT5514_SPI_BUF_LEN		240
 
-#define RT5514_BUFFER_VOICE_BASE	0x18001034
-#define RT5514_BUFFER_VOICE_LIMIT	0x18001038
-#define RT5514_BUFFER_VOICE_RP		0x1800103c
-#define RT5514_BUFFER_VOICE_SIZE	0x18001040
+#define RT5514_BUFFER_VOICE_BASE	0x18000200
+#define RT5514_BUFFER_VOICE_LIMIT	0x18000204
+#define RT5514_BUFFER_VOICE_WP		0x1800020c
 
 /* SPI Command */
 enum {
-- 
cgit v1.2.3


From 583ac685986963737997d0f6df569a3daab9c097 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 00:47:35 -0500
Subject: ASoC: tegra: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra30_i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index b2b279c96029..0b176ea24914 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -275,7 +275,7 @@ static int tegra30_i2s_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops tegra30_i2s_dai_ops = {
+static const struct snd_soc_dai_ops tegra30_i2s_dai_ops = {
 	.set_fmt	= tegra30_i2s_set_fmt,
 	.hw_params	= tegra30_i2s_hw_params,
 	.trigger	= tegra30_i2s_trigger,
-- 
cgit v1.2.3


From fe49cd980fc15cbb7eeb8d1a4ef45a14b807b55d Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 00:52:23 -0500
Subject: ASoC: sun8i-codec: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun8i-codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 5723c3404f6b..253ae0b5a4d4 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -341,7 +341,7 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = {
 	  "AIF1 Slot 0 Right"},
 };
 
-static struct snd_soc_dai_ops sun8i_codec_dai_ops = {
+static const struct snd_soc_dai_ops sun8i_codec_dai_ops = {
 	.hw_params = sun8i_codec_hw_params,
 	.set_fmt = sun8i_set_fmt,
 };
-- 
cgit v1.2.3


From 06305d72796076b09495daa4398a43ddb68a6caa Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 02:14:21 -0500
Subject: ASoC: fsl_spdif: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_spdif.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 1ff467c9598a..7e6cc4da0088 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -626,7 +626,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+static const struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 	.startup = fsl_spdif_startup,
 	.hw_params = fsl_spdif_hw_params,
 	.trigger = fsl_spdif_trigger,
-- 
cgit v1.2.3


From 5d29e95eec1451d35ef6ce5b851fbc8b04f82d39 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 02:19:25 -0500
Subject: ASoC: fsl_esai: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_esai.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 809a069d490b..cef79a1a620b 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -620,7 +620,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
-static struct snd_soc_dai_ops fsl_esai_dai_ops = {
+static const struct snd_soc_dai_ops fsl_esai_dai_ops = {
 	.startup = fsl_esai_startup,
 	.shutdown = fsl_esai_shutdown,
 	.trigger = fsl_esai_trigger,
-- 
cgit v1.2.3


From 29a22ebfa4c44bf96b3d00d90b5e3b8128d8a162 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 02:23:51 -0500
Subject: ASoC: fsl_asrc: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_asrc.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 8cfffa70c144..806d39927318 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -542,7 +542,7 @@ static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
-static struct snd_soc_dai_ops fsl_asrc_dai_ops = {
+static const struct snd_soc_dai_ops fsl_asrc_dai_ops = {
 	.hw_params    = fsl_asrc_dai_hw_params,
 	.hw_free      = fsl_asrc_dai_hw_free,
 	.trigger      = fsl_asrc_dai_trigger,
-- 
cgit v1.2.3


From 94372fceb2fb5d3d05d6db61c9658865d5e27ee4 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 02:08:53 -0500
Subject: ASoC: hisilicon: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/hisilicon/hi6210-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index b193d3beb253..0c8f86d4020e 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -517,7 +517,7 @@ static int hi6210_i2s_dai_probe(struct snd_soc_dai *dai)
 }
 
 
-static struct snd_soc_dai_ops hi6210_i2s_dai_ops = {
+static const struct snd_soc_dai_ops hi6210_i2s_dai_ops = {
 	.trigger	= hi6210_i2s_trigger,
 	.hw_params	= hi6210_i2s_hw_params,
 	.set_fmt	= hi6210_i2s_set_fmt,
-- 
cgit v1.2.3


From 549acff975c9245ea8d469c0defa50e52489227c Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 01:10:19 -0500
Subject: ASoC: mediatek: mt2701: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
index bc5d4db94de6..10dcd69c5ffa 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
@@ -595,7 +595,7 @@ static const struct snd_soc_dai_ops mt2701_afe_i2s_ops = {
 };
 
 /* MRG BE DAIs */
-static struct snd_soc_dai_ops mt2701_btmrg_ops = {
+static const struct snd_soc_dai_ops mt2701_btmrg_ops = {
 	.startup = mt2701_btmrg_startup,
 	.shutdown = mt2701_btmrg_shutdown,
 	.hw_params = mt2701_btmrg_hw_params,
-- 
cgit v1.2.3


From cb7534437956650ab0dcebf7a7b4795eb904d22a Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 01:05:08 -0500
Subject: ASoC: mmp-sspa: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/pxa/mmp-sspa.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index 9cc35012e6e5..64b85e30c1f8 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -380,7 +380,7 @@ static int mmp_sspa_probe(struct snd_soc_dai *dai)
 		SNDRV_PCM_FMTBIT_S24_LE | \
 		SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+static const struct snd_soc_dai_ops mmp_sspa_dai_ops = {
 	.startup	= mmp_sspa_startup,
 	.shutdown	= mmp_sspa_shutdown,
 	.trigger	= mmp_sspa_trigger,
-- 
cgit v1.2.3


From 45d059a33aa48cb19a5ff589f71abcc4de4a8fbf Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 00:22:09 -0500
Subject: ASoC: zx-i2s: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/zte/zx-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c
index 8bbad1d72bc5..9a0565937d1f 100644
--- a/sound/soc/zte/zx-i2s.c
+++ b/sound/soc/zte/zx-i2s.c
@@ -357,7 +357,7 @@ static void zx_i2s_shutdown(struct snd_pcm_substream *substream,
 	clk_disable_unprepare(zx_i2s->dai_pclk);
 }
 
-static struct snd_soc_dai_ops zx_i2s_dai_ops = {
+static const struct snd_soc_dai_ops zx_i2s_dai_ops = {
 	.trigger	= zx_i2s_trigger,
 	.hw_params	= zx_i2s_hw_params,
 	.set_fmt	= zx_i2s_set_fmt,
-- 
cgit v1.2.3


From 171dd51655b11331928dd39ff0e51a516e41424d Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 00:30:52 -0500
Subject: ASoC: zx-tdm: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/zte/zx-tdm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c
index bd632cc503b3..dc955272f58b 100644
--- a/sound/soc/zte/zx-tdm.c
+++ b/sound/soc/zte/zx-tdm.c
@@ -309,7 +309,7 @@ static void zx_tdm_shutdown(struct snd_pcm_substream *substream,
 	clk_disable_unprepare(zx_tdm->dai_wclk);
 }
 
-static struct snd_soc_dai_ops zx_tdm_dai_ops = {
+static const struct snd_soc_dai_ops zx_tdm_dai_ops = {
 	.trigger	= zx_tdm_trigger,
 	.hw_params	= zx_tdm_hw_params,
 	.set_fmt	= zx_tdm_set_fmt,
-- 
cgit v1.2.3


From 4d2c7e09a5c8ea3b20748c2942b6a686dced34ab Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 00:37:55 -0500
Subject: ASoC: zte: spdif: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/zte/zx-spdif.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c
index 9fa6463ce5d7..b143f9f682d2 100644
--- a/sound/soc/zte/zx-spdif.c
+++ b/sound/soc/zte/zx-spdif.c
@@ -264,7 +264,7 @@ static void zx_spdif_shutdown(struct snd_pcm_substream *substream,
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE \
 	| SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops zx_spdif_dai_ops = {
+static const struct snd_soc_dai_ops zx_spdif_dai_ops = {
 	.trigger	= zx_spdif_trigger,
 	.startup	= zx_spdif_startup,
 	.shutdown	= zx_spdif_shutdown,
-- 
cgit v1.2.3


From fde0543f1129967dfa73f1fa3477c7fe12052da5 Mon Sep 17 00:00:00 2001
From: Dan Murphy
Date: Wed, 12 Jul 2017 13:36:59 -0500
Subject: ASoC: tlv320aic32x4: Add support for tlv320aic32x6

Add support for the tlv320aic32x6 family of devices.
These devices share the same register set as the tlv320aic32x4
and this driver can be re-used.

TLV320aic32x6 data sheets are here:
tlv320aic3256 - http://www.ti.com/lit/gpn/tlv320aic3256
tlv320aic3206 - http://www.ti.com/lit/gpn/tlv320aic3206

Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt | 4 +++-
 sound/soc/codecs/tlv320aic32x4-i2c.c                      | 2 ++
 sound/soc/codecs/tlv320aic32x4-spi.c                      | 2 ++
 3 files changed, 7 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
index 5e2741af27be..9651fc7503a4 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -3,7 +3,9 @@ Texas Instruments - tlv320aic32x4 Codec module
 The tlv320aic32x4 serial control bus communicates through I2C protocols
 
 Required properties:
- - compatible: Should be "ti,tlv320aic32x4"
+ - compatible - "string" - One of:
+	"ti,tlv320aic32x4" TLV320AIC3204
+	"ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256
  - reg: I2C slave address
  - supply-*: Required supply regulators are:
     "iov" - digital IO power supply
diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c
index 59606cf3008f..385fa2e9525a 100644
--- a/sound/soc/codecs/tlv320aic32x4-i2c.c
+++ b/sound/soc/codecs/tlv320aic32x4-i2c.c
@@ -47,12 +47,14 @@ static int aic32x4_i2c_remove(struct i2c_client *i2c)
 
 static const struct i2c_device_id aic32x4_i2c_id[] = {
 	{ "tlv320aic32x4", 0 },
+	{ "tlv320aic32x6", 1 },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
 
 static const struct of_device_id aic32x4_of_id[] = {
 	{ .compatible = "ti,tlv320aic32x4", },
+	{ .compatible = "ti,tlv320aic32x6", },
 	{ /* senitel */ }
 };
 MODULE_DEVICE_TABLE(of, aic32x4_of_id);
diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c
index 724fcdd491b2..07d78ae51e05 100644
--- a/sound/soc/codecs/tlv320aic32x4-spi.c
+++ b/sound/soc/codecs/tlv320aic32x4-spi.c
@@ -48,12 +48,14 @@ static int aic32x4_spi_remove(struct spi_device *spi)
 
 static const struct spi_device_id aic32x4_spi_id[] = {
 	{ "tlv320aic32x4", 0 },
+	{ "tlv320aic32x6", 1 },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(spi, aic32x4_spi_id);
 
 static const struct of_device_id aic32x4_of_id[] = {
 	{ .compatible = "ti,tlv320aic32x4", },
+	{ .compatible = "ti,tlv320aic32x6", },
 	{ /* senitel */ }
 };
 MODULE_DEVICE_TABLE(of, aic32x4_of_id);
-- 
cgit v1.2.3


From b9045b9c6b79d2ea5b888a0f9e9aa156d09ba8a9 Mon Sep 17 00:00:00 2001
From: Dan Murphy
Date: Wed, 12 Jul 2017 13:37:00 -0500
Subject: ASoC: tlv320aic32x4: Add gpio configuration to the codec

Add the ability to configure the MFP1->MFP5 registers
as GPIOs.  In addition adding ALSA controls to get and set
the GPIO state.

Per the data sheet each MFP can be configured as a GPIO
input only, output only or either an input or output.

Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/tlv320aic32x4.txt    |   9 +
 include/sound/tlv320aic32x4.h                      |  23 +++
 sound/soc/codecs/tlv320aic32x4.c                   | 206 +++++++++++++++++++++
 sound/soc/codecs/tlv320aic32x4.h                   |   3 +
 4 files changed, 241 insertions(+)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
index 9651fc7503a4..ca75890f0d07 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -20,6 +20,8 @@ Optional properties:
  - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt
  - clocks/clock-names: Clock named 'mclk' for the master clock of the codec.
    See clock/clock-bindings.txt for information about the detailed format.
+ - aic32x4-gpio-func - <array of 5 int>
+	- Types are defined in include/sound/tlv320aic32x4.h
 
 
 Example:
@@ -29,4 +31,11 @@ codec: tlv320aic32x4@18 {
 	reg = <0x18>;
 	clocks = <&clks 201>;
 	clock-names = "mclk";
+	aic32x4-gpio-func= <
+			0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+			0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+			0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */
+			0xff /* AIC32X4_MFPX_DEFAULT_VALUE */
+			0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */
+		>;
 };
diff --git a/include/sound/tlv320aic32x4.h b/include/sound/tlv320aic32x4.h
index 24e5d991f148..22305c0ab31a 100644
--- a/include/sound/tlv320aic32x4.h
+++ b/include/sound/tlv320aic32x4.h
@@ -22,7 +22,30 @@
 #define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K	0x00000001
 #define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K	0x00000002
 
+/* GPIO API */
+#define AIC32X4_MFPX_DEFAULT_VALUE	0xff
+
+#define AIC32X4_MFP1_DIN_DISABLED	0
+#define AIC32X4_MFP1_DIN_ENABLED	0x2
+#define AIC32X4_MFP1_GPIO_IN		0x4
+
+#define AIC32X4_MFP2_GPIO_OUT_LOW	0x0
+#define AIC32X4_MFP2_GPIO_OUT_HIGH	0x1
+
+#define AIC32X4_MFP_GPIO_ENABLED	0x4
+
+#define AIC32X4_MFP5_GPIO_DISABLED	0x0
+#define AIC32X4_MFP5_GPIO_INPUT		0x8
+#define AIC32X4_MFP5_GPIO_OUTPUT	0xc
+#define AIC32X4_MFP5_GPIO_OUT_LOW	0x0
+#define AIC32X4_MFP5_GPIO_OUT_HIGH	0x1
+
+struct aic32x4_setup_data {
+	unsigned int gpio_func[5];
+};
+
 struct aic32x4_pdata {
+	struct aic32x4_setup_data *setup;
 	u32 power_cfg;
 	u32 micpga_routing;
 	bool swapdacs;
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 28fdfc5ec544..300ee2890df3 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -74,6 +74,152 @@ struct aic32x4_priv {
 	struct regulator *supply_iov;
 	struct regulator *supply_dv;
 	struct regulator *supply_av;
+
+	struct aic32x4_setup_data *setup;
+	struct device *dev;
+};
+
+static int aic32x4_get_mfp1_gpio(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u8 val;
+
+	val = snd_soc_read(codec, AIC32X4_DINCTL);
+
+	ucontrol->value.integer.value[0] = (val & 0x01);
+
+	return 0;
+};
+
+static int aic32x4_set_mfp2_gpio(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u8 val;
+	u8 gpio_check;
+
+	val = snd_soc_read(codec, AIC32X4_DOUTCTL);
+	gpio_check = (val & AIC32X4_MFP_GPIO_ENABLED);
+	if (gpio_check != AIC32X4_MFP_GPIO_ENABLED) {
+		printk(KERN_ERR "%s: MFP2 is not configure as a GPIO output\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	if (ucontrol->value.integer.value[0] == (val & AIC32X4_MFP2_GPIO_OUT_HIGH))
+		return 0;
+
+	if (ucontrol->value.integer.value[0])
+		val |= ucontrol->value.integer.value[0];
+	else
+		val &= ~AIC32X4_MFP2_GPIO_OUT_HIGH;
+
+	snd_soc_write(codec, AIC32X4_DOUTCTL, val);
+
+	return 0;
+};
+
+static int aic32x4_get_mfp3_gpio(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u8 val;
+
+	val = snd_soc_read(codec, AIC32X4_SCLKCTL);
+
+	ucontrol->value.integer.value[0] = (val & 0x01);
+
+	return 0;
+};
+
+static int aic32x4_set_mfp4_gpio(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u8 val;
+	u8 gpio_check;
+
+	val = snd_soc_read(codec, AIC32X4_MISOCTL);
+	gpio_check = (val & AIC32X4_MFP_GPIO_ENABLED);
+	if (gpio_check != AIC32X4_MFP_GPIO_ENABLED) {
+		printk(KERN_ERR "%s: MFP4 is not configure as a GPIO output\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	if (ucontrol->value.integer.value[0] == (val & AIC32X4_MFP5_GPIO_OUT_HIGH))
+		return 0;
+
+	if (ucontrol->value.integer.value[0])
+		val |= ucontrol->value.integer.value[0];
+	else
+		val &= ~AIC32X4_MFP5_GPIO_OUT_HIGH;
+
+	snd_soc_write(codec, AIC32X4_MISOCTL, val);
+
+	return 0;
+};
+
+static int aic32x4_get_mfp5_gpio(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u8 val;
+
+	val = snd_soc_read(codec, AIC32X4_GPIOCTL);
+	ucontrol->value.integer.value[0] = ((val & 0x2) >> 1);
+
+	return 0;
+};
+
+static int aic32x4_set_mfp5_gpio(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u8 val;
+	u8 gpio_check;
+
+	val = snd_soc_read(codec, AIC32X4_GPIOCTL);
+	gpio_check = (val & AIC32X4_MFP5_GPIO_OUTPUT);
+	if (gpio_check != AIC32X4_MFP5_GPIO_OUTPUT) {
+		printk(KERN_ERR "%s: MFP5 is not configure as a GPIO output\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	if (ucontrol->value.integer.value[0] == (val & 0x1))
+		return 0;
+
+	if (ucontrol->value.integer.value[0])
+		val |= ucontrol->value.integer.value[0];
+	else
+		val &= 0xfe;
+
+	snd_soc_write(codec, AIC32X4_GPIOCTL, val);
+
+	return 0;
+};
+
+static const struct snd_kcontrol_new aic32x4_mfp1[] = {
+	SOC_SINGLE_BOOL_EXT("MFP1 GPIO", 0, aic32x4_get_mfp1_gpio, NULL),
+};
+
+static const struct snd_kcontrol_new aic32x4_mfp2[] = {
+	SOC_SINGLE_BOOL_EXT("MFP2 GPIO", 0, NULL, aic32x4_set_mfp2_gpio),
+};
+
+static const struct snd_kcontrol_new aic32x4_mfp3[] = {
+	SOC_SINGLE_BOOL_EXT("MFP3 GPIO", 0, aic32x4_get_mfp3_gpio, NULL),
+};
+
+static const struct snd_kcontrol_new aic32x4_mfp4[] = {
+	SOC_SINGLE_BOOL_EXT("MFP4 GPIO", 0, NULL, aic32x4_set_mfp4_gpio),
+};
+
+static const struct snd_kcontrol_new aic32x4_mfp5[] = {
+	SOC_SINGLE_BOOL_EXT("MFP5 GPIO", 0, aic32x4_get_mfp5_gpio,
+		aic32x4_set_mfp5_gpio),
 };
 
 /* 0dB min, 0.5dB steps */
@@ -734,6 +880,52 @@ static struct snd_soc_dai_driver aic32x4_dai = {
 	.symmetric_rates = 1,
 };
 
+static void aic32x4_setup_gpios(struct snd_soc_codec *codec)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+
+	/* setup GPIO functions */
+	/* MFP1 */
+	if (aic32x4->setup->gpio_func[0] != AIC32X4_MFPX_DEFAULT_VALUE) {
+		snd_soc_write(codec, AIC32X4_DINCTL,
+		      aic32x4->setup->gpio_func[0]);
+		snd_soc_add_codec_controls(codec, aic32x4_mfp1,
+			ARRAY_SIZE(aic32x4_mfp1));
+	}
+
+	/* MFP2 */
+	if (aic32x4->setup->gpio_func[1] != AIC32X4_MFPX_DEFAULT_VALUE) {
+		snd_soc_write(codec, AIC32X4_DOUTCTL,
+		      aic32x4->setup->gpio_func[1]);
+		snd_soc_add_codec_controls(codec, aic32x4_mfp2,
+			ARRAY_SIZE(aic32x4_mfp2));
+	}
+
+	/* MFP3 */
+	if (aic32x4->setup->gpio_func[2] != AIC32X4_MFPX_DEFAULT_VALUE) {
+		snd_soc_write(codec, AIC32X4_SCLKCTL,
+		      aic32x4->setup->gpio_func[2]);
+		snd_soc_add_codec_controls(codec, aic32x4_mfp3,
+			ARRAY_SIZE(aic32x4_mfp3));
+	}
+
+	/* MFP4 */
+	if (aic32x4->setup->gpio_func[3] != AIC32X4_MFPX_DEFAULT_VALUE) {
+		snd_soc_write(codec, AIC32X4_MISOCTL,
+		      aic32x4->setup->gpio_func[3]);
+		snd_soc_add_codec_controls(codec, aic32x4_mfp4,
+			ARRAY_SIZE(aic32x4_mfp4));
+	}
+
+	/* MFP5 */
+	if (aic32x4->setup->gpio_func[4] != AIC32X4_MFPX_DEFAULT_VALUE) {
+		snd_soc_write(codec, AIC32X4_GPIOCTL,
+		      aic32x4->setup->gpio_func[4]);
+		snd_soc_add_codec_controls(codec, aic32x4_mfp5,
+			ARRAY_SIZE(aic32x4_mfp5));
+	}
+}
+
 static int aic32x4_codec_probe(struct snd_soc_codec *codec)
 {
 	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
@@ -746,6 +938,9 @@ static int aic32x4_codec_probe(struct snd_soc_codec *codec)
 
 	snd_soc_write(codec, AIC32X4_RESET, 0x01);
 
+	if (aic32x4->setup)
+		aic32x4_setup_gpios(codec);
+
 	/* Power platform configuration */
 	if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) {
 		snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN |
@@ -810,10 +1005,20 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4,
 		struct device_node *np)
 {
+	struct aic32x4_setup_data *aic32x4_setup;
+
+	aic32x4_setup = devm_kzalloc(aic32x4->dev, sizeof(*aic32x4_setup),
+							GFP_KERNEL);
+	if (!aic32x4_setup)
+		return -ENOMEM;
+
 	aic32x4->swapdacs = false;
 	aic32x4->micpga_routing = 0;
 	aic32x4->rstn_gpio = of_get_named_gpio(np, "reset-gpios", 0);
 
+	if (of_property_read_u32_array(np, "aic32x4-gpio-func",
+				aic32x4_setup->gpio_func, 5) >= 0)
+		aic32x4->setup = aic32x4_setup;
 	return 0;
 }
 
@@ -932,6 +1137,7 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap)
 	if (aic32x4 == NULL)
 		return -ENOMEM;
 
+	aic32x4->dev = dev;
 	dev_set_drvdata(dev, aic32x4);
 
 	if (pdata) {
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index a197dd51addc..da7cec482bcb 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -44,8 +44,11 @@ int aic32x4_remove(struct device *dev);
 #define AIC32X4_IFACE4		31
 #define AIC32X4_IFACE5		32
 #define AIC32X4_IFACE6		33
+#define AIC32X4_GPIOCTL		52
 #define AIC32X4_DOUTCTL		53
 #define AIC32X4_DINCTL		54
+#define AIC32X4_MISOCTL		55
+#define AIC32X4_SCLKCTL		56
 #define AIC32X4_DACSPB		60
 #define AIC32X4_ADCSPB		61
 #define AIC32X4_DACSETUP	63
-- 
cgit v1.2.3


From 72a63c6e78887f5c2063ce147b57fdfecf366e0e Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:37:26 -0500
Subject: ASoC: cs42l42: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 55caf5ff3e64..a2324a0e72ee 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -911,7 +911,7 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute)
 			SNDRV_PCM_FMTBIT_S32_LE)
 
 
-static struct snd_soc_dai_ops cs42l42_ops = {
+static const struct snd_soc_dai_ops cs42l42_ops = {
 	.hw_params	= cs42l42_pcm_hw_params,
 	.set_fmt	= cs42l42_set_dai_fmt,
 	.set_sysclk	= cs42l42_set_sysclk,
-- 
cgit v1.2.3


From 6ccf3a6d031c47d6319cb56ee35dd4b546905e91 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:37:36 -0500
Subject: ASoC: rt5663: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 963cda9c995a..71c073c63866 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -2888,7 +2888,7 @@ static int rt5663_resume(struct snd_soc_codec *codec)
 #define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-static struct snd_soc_dai_ops rt5663_aif_dai_ops = {
+static const struct snd_soc_dai_ops rt5663_aif_dai_ops = {
 	.hw_params = rt5663_hw_params,
 	.set_fmt = rt5663_set_dai_fmt,
 	.set_sysclk = rt5663_set_dai_sysclk,
-- 
cgit v1.2.3


From f1d68ea36f4fe97ed1de8d40eaa43758b8386c29 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:37:43 -0500
Subject: ASoC: rt5616: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5616.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c
index 7d6e0823f98f..33bdfe7c394a 100644
--- a/sound/soc/codecs/rt5616.c
+++ b/sound/soc/codecs/rt5616.c
@@ -1265,7 +1265,7 @@ static int rt5616_resume(struct snd_soc_codec *codec)
 #define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-static struct snd_soc_dai_ops rt5616_aif_dai_ops = {
+static const struct snd_soc_dai_ops rt5616_aif_dai_ops = {
 	.hw_params = rt5616_hw_params,
 	.set_fmt = rt5616_set_dai_fmt,
 	.set_sysclk = rt5616_set_dai_sysclk,
-- 
cgit v1.2.3


From af2f010e0f12fdeb26fce502adf4b353daeb302c Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:37:48 -0500
Subject: ASoC: msm8916-wcd-analog: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-analog.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index a78802920c3c..aec1e1626993 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -753,7 +753,7 @@ static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream,
 			    RST_CTL_DIG_SW_RST_N_MASK, 0);
 }
 
-static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = {
+static const struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = {
 	.startup = pm8916_wcd_analog_startup,
 	.shutdown = pm8916_wcd_analog_shutdown,
 };
-- 
cgit v1.2.3


From dfcfd4b68a551b7038e046040dac2a467f5efd10 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:37:54 -0500
Subject: ASoC: max98926: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98926.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
index 1eff7e0b092e..f16145d8ee6b 100644
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -459,7 +459,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream,
 #define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops max98926_dai_ops = {
+static const struct snd_soc_dai_ops max98926_dai_ops = {
 	.set_fmt = max98926_dai_set_fmt,
 	.hw_params = max98926_dai_hw_params,
 };
-- 
cgit v1.2.3


From 9894dbafb6f864207f9e6aeaca2c8de949955651 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:37:59 -0500
Subject: ASoC: rt5514: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Also, make rt5514_aif_dai_ops static, as it is not used outside this
module.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index a71e63097aaf..4ca73bf83cc1 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -1006,7 +1006,7 @@ static int rt5514_i2c_write(void *context, unsigned int reg, unsigned int val)
 #define RT5514_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-struct snd_soc_dai_ops rt5514_aif_dai_ops = {
+static const struct snd_soc_dai_ops rt5514_aif_dai_ops = {
 	.hw_params = rt5514_hw_params,
 	.set_fmt = rt5514_set_dai_fmt,
 	.set_sysclk = rt5514_set_dai_sysclk,
-- 
cgit v1.2.3


From a9689bb890ce633c3015ed51e4e2c6ee320fe4dc Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:38:14 -0500
Subject: ASoC: codecs: msm8916-wcd-digital: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-digital.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index f690442af8c9..7e3794fb8c2c 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -788,7 +788,7 @@ static void msm8916_wcd_digital_shutdown(struct snd_pcm_substream *substream,
 			    LPASS_CDC_CLK_PDM_CTL_PDM_CLK_SEL_MASK, 0);
 }
 
-static struct snd_soc_dai_ops msm8916_wcd_digital_dai_ops = {
+static const struct snd_soc_dai_ops msm8916_wcd_digital_dai_ops = {
 	.startup = msm8916_wcd_digital_startup,
 	.shutdown = msm8916_wcd_digital_shutdown,
 	.hw_params = msm8916_wcd_digital_hw_params,
-- 
cgit v1.2.3


From df9401b47c346480e1c1068f12c1cd32e45809f1 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:38:26 -0500
Subject: ASoC: max9867: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max9867.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index 2a40a69a7513..3e421e9d14a5 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -350,7 +350,7 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static struct snd_soc_dai_ops max9867_dai_ops = {
+static const struct snd_soc_dai_ops max9867_dai_ops = {
 	.set_fmt = max9867_dai_set_fmt,
 	.set_sysclk	= max9867_set_dai_sysclk,
 	.prepare	= max9867_prepare,
-- 
cgit v1.2.3


From fcf005a645e6855d68f98b9455f6b47ac2ed161c Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:38:32 -0500
Subject: ASoC: rk3036: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/inno_rk3036.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c
index b918ba5c8ce5..edf42d252fa1 100644
--- a/sound/soc/codecs/inno_rk3036.c
+++ b/sound/soc/codecs/inno_rk3036.c
@@ -310,7 +310,7 @@ static int rk3036_codec_dai_hw_params(struct snd_pcm_substream *substream,
 			   SNDRV_PCM_FMTBIT_S24_LE  | \
 			   SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops rk3036_codec_dai_ops = {
+static const struct snd_soc_dai_ops rk3036_codec_dai_ops = {
 	.set_fmt	= rk3036_codec_dai_set_fmt,
 	.hw_params	= rk3036_codec_dai_hw_params,
 };
-- 
cgit v1.2.3


From 61b3b3cc68c4161ba202f401a6572250360ff18c Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:38:38 -0500
Subject: ASoC: hdac_hdmi: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdac_hdmi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index bc2e74ff3b2d..e6de50acefd4 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1360,7 +1360,7 @@ static void hdac_hdmi_skl_enable_dp12(struct hdac_device *hdac)
 
 }
 
-static struct snd_soc_dai_ops hdmi_dai_ops = {
+static const struct snd_soc_dai_ops hdmi_dai_ops = {
 	.startup = hdac_hdmi_pcm_open,
 	.shutdown = hdac_hdmi_pcm_close,
 	.hw_params = hdac_hdmi_set_hw_params,
-- 
cgit v1.2.3


From 73a995ca90544914a97ed33a4f7baef44ad110d0 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:38:43 -0500
Subject: ASoC: dwc: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/dwc/dwc-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index 916067638180..eebe9d34e7a0 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -381,7 +381,7 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 	return ret;
 }
 
-static struct snd_soc_dai_ops dw_i2s_dai_ops = {
+static const struct snd_soc_dai_ops dw_i2s_dai_ops = {
 	.startup	= dw_i2s_startup,
 	.shutdown	= dw_i2s_shutdown,
 	.hw_params	= dw_i2s_hw_params,
-- 
cgit v1.2.3


From 86464ef83af934b7e489c4e100d0136319794010 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Thu, 13 Jul 2017 15:38:47 -0500
Subject: ASoC: tas5720: constify snd_soc_dai_ops structure

This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.

Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas5720.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index c65b917598d2..01e9ed242378 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -507,7 +507,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
 #define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\
 			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
+static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
 	.hw_params	= tas5720_hw_params,
 	.set_fmt	= tas5720_set_dai_fmt,
 	.set_tdm_slot	= tas5720_set_dai_tdm_slot,
-- 
cgit v1.2.3


From 32e1b60d8f91be932c4a624d4153224895b65ec7 Mon Sep 17 00:00:00 2001
From: Gustavo A. R. Silva
Date: Mon, 17 Jul 2017 10:30:20 -0500
Subject: ASoC: rsnd: remove unnecessary static in rsnd_ssiu_probe()

Remove unnecessary static on local variable ops.
Such variable is initialized before being used,
on every execution path throughout the function.
The static has no benefit and, removing it reduces
the code size.

This issue was detected using Coccinelle and the following semantic patch:

@bad exists@
position p;
identifier x;
type T;
@@

static T x@p;
...
x = <+...x...+>

@@
identifier x;
expression e;
type T;
position p != bad.p;
@@

-static
 T x@p;
 ... when != x
     when strict
?x = e;

In the following log you can see the difference in the code size. Also,
there is a significant difference in the bss segment. This log is the
output of the size command, before and after the code change:

before:
   text    data     bss     dec     hex filename
   3211     680      64    3955     f73 sound/soc/sh/rcar/ssiu.o

after:
   text    data     bss     dec     hex filename
   3207     592       0    3799     ed7 sound/soc/sh/rcar/ssiu.o

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssiu.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index bed2c9c0004b..4d948757d300 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -250,7 +250,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv)
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct rsnd_ssiu *ssiu;
-	static struct rsnd_mod_ops *ops;
+	struct rsnd_mod_ops *ops;
 	int i, nr, ret;
 
 	/* same number to SSI */
-- 
cgit v1.2.3


From d26ed933581d924937aa92251afe1751cf86661c Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Tue, 18 Jul 2017 10:12:58 +0800
Subject: ASoC: rt5663: Modify the default value for ASRC function

The patch modifies the default value for ASRC function. It could prevent
the pop noise and recording no sound with ASRC function.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 71c073c63866..d3f7bc2173e8 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -60,6 +60,7 @@ struct rt5663_priv {
 
 static const struct reg_sequence rt5663_patch_list[] = {
 	{ 0x002a, 0x8080 },
+	{ 0x0086, 0x0028 },
 };
 
 static const struct reg_default rt5663_v2_reg[] = {
@@ -509,7 +510,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0082, 0x0000 },
 	{ 0x0083, 0x0000 },
 	{ 0x0084, 0x0000 },
-	{ 0x0086, 0x0008 },
+	{ 0x0086, 0x0028 },
 	{ 0x0087, 0x0000 },
 	{ 0x008a, 0x0000 },
 	{ 0x008b, 0x0000 },
-- 
cgit v1.2.3


From c7e79b2b2d2d7c13c6162ed8672357f4ad0fd8f5 Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Fri, 14 Jul 2017 14:25:52 +0800
Subject: ASoC: rt274: add rt274 codec driver

RT274 is a HD-A/SOC dual mode codec. This is the initial codec driver
of SOC mode.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/rt274.txt |   33 +
 sound/soc/codecs/Kconfig                          |    7 +
 sound/soc/codecs/Makefile                         |    2 +
 sound/soc/codecs/rt274.c                          | 1228 +++++++++++++++++++++
 sound/soc/codecs/rt274.h                          |  217 ++++
 5 files changed, 1487 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/rt274.txt
 create mode 100644 sound/soc/codecs/rt274.c
 create mode 100644 sound/soc/codecs/rt274.h

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt
new file mode 100644
index 000000000000..e9a6178c78cf
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt274.txt
@@ -0,0 +1,33 @@
+RT274 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt274".
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- interrupts : The CODEC's interrupt output.
+
+
+Pins on the device (for linking into audio routes) for RT274:
+
+  * DMIC1 Pin
+  * DMIC2 Pin
+  * MIC
+  * LINE1
+  * LINE2
+  * HPO Pin
+  * SPDIF
+  * LINE3
+
+Example:
+
+codec: rt274@1c {
+	compatible = "realtek,rt274";
+	reg = <0x1c>;
+	interrupts = <7 IRQ_TYPE_EDGE_FALLING>;
+};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 6c78b0b49b81..024ddc9938ed 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -114,6 +114,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM5102A
 	select SND_SOC_PCM512x_I2C if I2C
 	select SND_SOC_PCM512x_SPI if SPI_MASTER
+	select SND_SOC_RT274 if I2C
 	select SND_SOC_RT286 if I2C
 	select SND_SOC_RT298 if I2C
 	select SND_SOC_RT5514 if I2C
@@ -716,11 +717,17 @@ config SND_SOC_RL6231
 
 config SND_SOC_RL6347A
 	tristate
+	default y if SND_SOC_RT274=y
 	default y if SND_SOC_RT286=y
 	default y if SND_SOC_RT298=y
+	default m if SND_SOC_RT274=m
 	default m if SND_SOC_RT286=m
 	default m if SND_SOC_RT298=m
 
+config SND_SOC_RT274
+	tristate
+	depends on I2C
+
 config SND_SOC_RT286
 	tristate
 	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1755a54e3dc9..a8a4b0797f46 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -113,6 +113,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
 snd-soc-pcm512x-spi-objs := pcm512x-spi.o
 snd-soc-rl6231-objs := rl6231.o
 snd-soc-rl6347a-objs := rl6347a.o
+snd-soc-rt274-objs := rt274.o
 snd-soc-rt286-objs := rt286.o
 snd-soc-rt298-objs := rt298.o
 snd-soc-rt5514-objs := rt5514.o
@@ -349,6 +350,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C)	+= snd-soc-pcm512x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM512x_SPI)	+= snd-soc-pcm512x-spi.o
 obj-$(CONFIG_SND_SOC_RL6231)	+= snd-soc-rl6231.o
 obj-$(CONFIG_SND_SOC_RL6347A)	+= snd-soc-rl6347a.o
+obj-$(CONFIG_SND_SOC_RT274)	+= snd-soc-rt274.o
 obj-$(CONFIG_SND_SOC_RT286)	+= snd-soc-rt286.o
 obj-$(CONFIG_SND_SOC_RT298)	+= snd-soc-rt298.o
 obj-$(CONFIG_SND_SOC_RT5514)	+= snd-soc-rt5514.o
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
new file mode 100644
index 000000000000..81e29ba2f50c
--- /dev/null
+++ b/sound/soc/codecs/rt274.c
@@ -0,0 +1,1228 @@
+/*
+ * rt274.c  --  RT274 ALSA SoC audio codec driver
+ *
+ * Copyright 2017 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <linux/dmi.h>
+#include <linux/acpi.h>
+#include <sound/core.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 <sound/jack.h>
+#include <linux/workqueue.h>
+
+#include "rl6347a.h"
+#include "rt274.h"
+
+#define RT274_VENDOR_ID 0x10ec0274
+
+struct rt274_priv {
+	struct reg_default *index_cache;
+	int index_cache_size;
+	struct regmap *regmap;
+	struct snd_soc_codec *codec;
+	struct i2c_client *i2c;
+	struct snd_soc_jack *jack;
+	struct delayed_work jack_detect_work;
+	int sys_clk;
+	int clk_id;
+	int fs;
+	bool master;
+};
+
+static const struct reg_default rt274_index_def[] = {
+	{ 0x00, 0x1004 },
+	{ 0x01, 0xaaaa },
+	{ 0x02, 0x88aa },
+	{ 0x03, 0x0002 },
+	{ 0x04, 0xaa09 },
+	{ 0x05, 0x0700 },
+	{ 0x06, 0x6110 },
+	{ 0x07, 0x0200 },
+	{ 0x08, 0xa807 },
+	{ 0x09, 0x0021 },
+	{ 0x0a, 0x7770 },
+	{ 0x0b, 0x7770 },
+	{ 0x0c, 0x002b },
+	{ 0x0d, 0x2420 },
+	{ 0x0e, 0x65c0 },
+	{ 0x0f, 0x7770 },
+	{ 0x10, 0x0420 },
+	{ 0x11, 0x7418 },
+	{ 0x12, 0x6bd0 },
+	{ 0x13, 0x645f },
+	{ 0x14, 0x0400 },
+	{ 0x15, 0x8ccc },
+	{ 0x16, 0x4c50 },
+	{ 0x17, 0xff00 },
+	{ 0x18, 0x0003 },
+	{ 0x19, 0x2c11 },
+	{ 0x1a, 0x830b },
+	{ 0x1b, 0x4e4b },
+	{ 0x1c, 0x0000 },
+	{ 0x1d, 0x0000 },
+	{ 0x1e, 0x0000 },
+	{ 0x1f, 0x0000 },
+	{ 0x20, 0x51ff },
+	{ 0x21, 0x8000 },
+	{ 0x22, 0x8f00 },
+	{ 0x23, 0x88f4 },
+	{ 0x24, 0x0000 },
+	{ 0x25, 0x0000 },
+	{ 0x26, 0x0000 },
+	{ 0x27, 0x0000 },
+	{ 0x28, 0x0000 },
+	{ 0x29, 0x3000 },
+	{ 0x2a, 0x0000 },
+	{ 0x2b, 0x0000 },
+	{ 0x2c, 0x0f00 },
+	{ 0x2d, 0x100f },
+	{ 0x2e, 0x2902 },
+	{ 0x2f, 0xe280 },
+	{ 0x30, 0x1000 },
+	{ 0x31, 0x8400 },
+	{ 0x32, 0x5aaa },
+	{ 0x33, 0x8420 },
+	{ 0x34, 0xa20c },
+	{ 0x35, 0x096a },
+	{ 0x36, 0x5757 },
+	{ 0x37, 0xfe05 },
+	{ 0x38, 0x4901 },
+	{ 0x39, 0x110a },
+	{ 0x3a, 0x0010 },
+	{ 0x3b, 0x60d9 },
+	{ 0x3c, 0xf214 },
+	{ 0x3d, 0xc2ba },
+	{ 0x3e, 0xa928 },
+	{ 0x3f, 0x0000 },
+	{ 0x40, 0x9800 },
+	{ 0x41, 0x0000 },
+	{ 0x42, 0x2000 },
+	{ 0x43, 0x3d90 },
+	{ 0x44, 0x4900 },
+	{ 0x45, 0x5289 },
+	{ 0x46, 0x0004 },
+	{ 0x47, 0xa47a },
+	{ 0x48, 0xd049 },
+	{ 0x49, 0x0049 },
+	{ 0x4a, 0xa83b },
+	{ 0x4b, 0x0777 },
+	{ 0x4c, 0x065c },
+	{ 0x4d, 0x7fff },
+	{ 0x4e, 0x7fff },
+	{ 0x4f, 0x0000 },
+	{ 0x50, 0x0000 },
+	{ 0x51, 0x0000 },
+	{ 0x52, 0xbf5f },
+	{ 0x53, 0x3320 },
+	{ 0x54, 0xcc00 },
+	{ 0x55, 0x0000 },
+	{ 0x56, 0x3f00 },
+	{ 0x57, 0x0000 },
+	{ 0x58, 0x0000 },
+	{ 0x59, 0x0000 },
+	{ 0x5a, 0x1300 },
+	{ 0x5b, 0x005f },
+	{ 0x5c, 0x0000 },
+	{ 0x5d, 0x1001 },
+	{ 0x5e, 0x1000 },
+	{ 0x5f, 0x0000 },
+	{ 0x60, 0x5554 },
+	{ 0x61, 0xffc0 },
+	{ 0x62, 0xa000 },
+	{ 0x63, 0xd010 },
+	{ 0x64, 0x0000 },
+	{ 0x65, 0x3fb1 },
+	{ 0x66, 0x1881 },
+	{ 0x67, 0xc810 },
+	{ 0x68, 0x2000 },
+	{ 0x69, 0xfff0 },
+	{ 0x6a, 0x0300 },
+	{ 0x6b, 0x5060 },
+	{ 0x6c, 0x0000 },
+	{ 0x6d, 0x0000 },
+	{ 0x6e, 0x0c25 },
+	{ 0x6f, 0x0c0b },
+	{ 0x70, 0x8000 },
+	{ 0x71, 0x4008 },
+	{ 0x72, 0x0000 },
+	{ 0x73, 0x0800 },
+	{ 0x74, 0xa28f },
+	{ 0x75, 0xa050 },
+	{ 0x76, 0x7fe8 },
+	{ 0x77, 0xdb8c },
+	{ 0x78, 0x0000 },
+	{ 0x79, 0x0000 },
+	{ 0x7a, 0x2a96 },
+	{ 0x7b, 0x800f },
+	{ 0x7c, 0x0200 },
+	{ 0x7d, 0x1600 },
+	{ 0x7e, 0x0000 },
+	{ 0x7f, 0x0000 },
+};
+#define INDEX_CACHE_SIZE ARRAY_SIZE(rt274_index_def)
+
+static const struct reg_default rt274_reg[] = {
+	{ 0x00170500, 0x00000400 },
+	{ 0x00220000, 0x00000031 },
+	{ 0x00239000, 0x00000057 },
+	{ 0x0023a000, 0x00000057 },
+	{ 0x00270500, 0x00000400 },
+	{ 0x00370500, 0x00000400 },
+	{ 0x00870500, 0x00000400 },
+	{ 0x00920000, 0x00000031 },
+	{ 0x00935000, 0x00000097 },
+	{ 0x00936000, 0x00000097 },
+	{ 0x00970500, 0x00000400 },
+	{ 0x00b37000, 0x00000400 },
+	{ 0x00b37200, 0x00000400 },
+	{ 0x00b37300, 0x00000400 },
+	{ 0x00c37000, 0x00000400 },
+	{ 0x00c37100, 0x00000400 },
+	{ 0x01270500, 0x00000400 },
+	{ 0x01370500, 0x00000400 },
+	{ 0x01371f00, 0x411111f0 },
+	{ 0x01937000, 0x00000000 },
+	{ 0x01970500, 0x00000400 },
+	{ 0x02050000, 0x0000001b },
+	{ 0x02139000, 0x00000080 },
+	{ 0x0213a000, 0x00000080 },
+	{ 0x02170100, 0x00000001 },
+	{ 0x02170500, 0x00000400 },
+	{ 0x02170700, 0x00000000 },
+	{ 0x02270100, 0x00000000 },
+	{ 0x02370100, 0x00000000 },
+	{ 0x01970700, 0x00000020 },
+	{ 0x00830000, 0x00000097 },
+	{ 0x00930000, 0x00000097 },
+	{ 0x01270700, 0x00000000 },
+};
+
+static bool rt274_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case 0 ... 0xff:
+	case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+	case RT274_GET_HP_SENSE:
+	case RT274_GET_MIC_SENSE:
+	case RT274_PROC_COEF:
+	case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0):
+	case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0):
+		return true;
+	default:
+		return false;
+	}
+
+
+}
+
+static bool rt274_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case 0 ... 0xff:
+	case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+	case RT274_GET_HP_SENSE:
+	case RT274_GET_MIC_SENSE:
+	case RT274_SET_AUDIO_POWER:
+	case RT274_SET_HPO_POWER:
+	case RT274_SET_DMIC1_POWER:
+	case RT274_LOUT_MUX:
+	case RT274_HPO_MUX:
+	case RT274_ADC0_MUX:
+	case RT274_ADC1_MUX:
+	case RT274_SET_MIC:
+	case RT274_SET_PIN_HPO:
+	case RT274_SET_PIN_LOUT3:
+	case RT274_SET_PIN_DMIC1:
+	case RT274_SET_AMP_GAIN_HPO:
+	case RT274_SET_DMIC2_DEFAULT:
+	case RT274_DAC0L_GAIN:
+	case RT274_DAC0R_GAIN:
+	case RT274_DAC1L_GAIN:
+	case RT274_DAC1R_GAIN:
+	case RT274_ADCL_GAIN:
+	case RT274_ADCR_GAIN:
+	case RT274_MIC_GAIN:
+	case RT274_HPOL_GAIN:
+	case RT274_HPOR_GAIN:
+	case RT274_LOUTL_GAIN:
+	case RT274_LOUTR_GAIN:
+	case RT274_DAC_FORMAT:
+	case RT274_ADC_FORMAT:
+	case RT274_COEF_INDEX:
+	case RT274_PROC_COEF:
+	case RT274_SET_AMP_GAIN_ADC_IN1:
+	case RT274_SET_AMP_GAIN_ADC_IN2:
+	case RT274_SET_POWER(RT274_DAC_OUT0):
+	case RT274_SET_POWER(RT274_DAC_OUT1):
+	case RT274_SET_POWER(RT274_ADC_IN1):
+	case RT274_SET_POWER(RT274_ADC_IN2):
+	case RT274_SET_POWER(RT274_DMIC2):
+	case RT274_SET_POWER(RT274_MIC):
+	case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0):
+	case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0):
+	case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0):
+	case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0):
+	case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0):
+	case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0):
+	case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0):
+		return true;
+	default:
+		return false;
+	}
+}
+
+#ifdef CONFIG_PM
+static void rt274_index_sync(struct snd_soc_codec *codec)
+{
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	for (i = 0; i < INDEX_CACHE_SIZE; i++) {
+		snd_soc_write(codec, rt274->index_cache[i].reg,
+				  rt274->index_cache[i].def);
+	}
+}
+#endif
+
+static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic)
+{
+	unsigned int buf;
+
+	*hp = false;
+	*mic = false;
+
+	if (!rt274->codec)
+		return -EINVAL;
+
+	regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf);
+	*hp = buf & 0x80000000;
+	regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf);
+	*mic = buf & 0x80000000;
+
+	pr_debug("*hp = %d *mic = %d\n", *hp, *mic);
+
+	return 0;
+}
+
+static void rt274_jack_detect_work(struct work_struct *work)
+{
+	struct rt274_priv *rt274 =
+		container_of(work, struct rt274_priv, jack_detect_work.work);
+	int status = 0;
+	bool hp = false;
+	bool mic = false;
+
+	if (rt274_jack_detect(rt274, &hp, &mic) < 0)
+		return;
+
+	if (hp == true)
+		status |= SND_JACK_HEADPHONE;
+
+	if (mic == true)
+		status |= SND_JACK_MICROPHONE;
+
+	snd_soc_jack_report(rt274->jack, status,
+		SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+}
+
+static irqreturn_t rt274_irq(int irq, void *data);
+
+static int rt274_mic_detect(struct snd_soc_codec *codec,
+	struct snd_soc_jack *jack,  void *data)
+{
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	if (jack == NULL) {
+		/* Disable jack detection */
+		regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
+					RT274_IRQ_EN, RT274_IRQ_DIS);
+
+		return 0;
+	}
+	rt274->jack = jack;
+
+	regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
+				RT274_IRQ_EN, RT274_IRQ_EN);
+
+	/* Send an initial report */
+	rt274_irq(0, rt274);
+
+	return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0);
+
+static const struct snd_kcontrol_new rt274_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT274_DAC0L_GAIN,
+			 RT274_DAC0R_GAIN, 0, 0x7f, 0, out_vol_tlv),
+	SOC_DOUBLE_R_TLV("DAC1 Playback Volume", RT274_DAC1L_GAIN,
+			 RT274_DAC1R_GAIN, 0, 0x7f, 0, out_vol_tlv),
+	SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT274_ADCL_GAIN,
+			    RT274_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+	SOC_DOUBLE_R("ADC0 Capture Switch", RT274_ADCL_GAIN,
+			    RT274_ADCR_GAIN, RT274_MUTE_SFT, 1, 1),
+	SOC_SINGLE_TLV("AMIC Volume", RT274_MIC_GAIN,
+			    0, 0x3, 0, mic_vol_tlv),
+};
+
+static const struct snd_kcontrol_new hpol_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOL_GAIN,
+			RT274_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hpor_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOR_GAIN,
+			RT274_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new loutl_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTL_GAIN,
+			RT274_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new loutr_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTR_GAIN,
+			RT274_MUTE_SFT, 1, 1);
+
+/* ADC0 source */
+static const char * const rt274_adc_src[] = {
+	"Mic", "Line1", "Line2", "Dmic"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	rt274_adc0_enum, RT274_ADC0_MUX, RT274_ADC_SEL_SFT,
+	rt274_adc_src);
+
+static const struct snd_kcontrol_new rt274_adc0_mux =
+	SOC_DAPM_ENUM("ADC 0 source", rt274_adc0_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+	rt274_adc1_enum, RT274_ADC1_MUX, RT274_ADC_SEL_SFT,
+	rt274_adc_src);
+
+static const struct snd_kcontrol_new rt274_adc1_mux =
+	SOC_DAPM_ENUM("ADC 1 source", rt274_adc1_enum);
+
+static const char * const rt274_dac_src[] = {
+	"DAC OUT0", "DAC OUT1"
+};
+/* HP-OUT source */
+static SOC_ENUM_SINGLE_DECL(rt274_hpo_enum, RT274_HPO_MUX,
+				0, rt274_dac_src);
+
+static const struct snd_kcontrol_new rt274_hpo_mux =
+SOC_DAPM_ENUM("HPO source", rt274_hpo_enum);
+
+/* Line out source */
+static SOC_ENUM_SINGLE_DECL(rt274_lout_enum, RT274_LOUT_MUX,
+				0, rt274_dac_src);
+
+static const struct snd_kcontrol_new rt274_lout_mux =
+SOC_DAPM_ENUM("LOUT source", rt274_lout_enum);
+
+static const struct snd_soc_dapm_widget rt274_dapm_widgets[] = {
+	/* Input Lines */
+	SND_SOC_DAPM_INPUT("DMIC1 Pin"),
+	SND_SOC_DAPM_INPUT("DMIC2 Pin"),
+	SND_SOC_DAPM_INPUT("MIC"),
+	SND_SOC_DAPM_INPUT("LINE1"),
+	SND_SOC_DAPM_INPUT("LINE2"),
+
+	/* DMIC */
+	SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC 0", NULL, RT274_SET_STREAMID_ADC1, 4, 0),
+	SND_SOC_DAPM_ADC("ADC 1", NULL, RT274_SET_STREAMID_ADC2, 4, 0),
+
+	/* ADC Mux */
+	SND_SOC_DAPM_MUX("ADC 0 Mux", SND_SOC_NOPM, 0, 0,
+		&rt274_adc0_mux),
+	SND_SOC_DAPM_MUX("ADC 1 Mux", SND_SOC_NOPM, 0, 0,
+		&rt274_adc1_mux),
+
+	/* Audio Interface */
+	SND_SOC_DAPM_AIF_IN("AIF1RXL", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF1RXR", "AIF1 Playback", 1, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF1TXL", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF1TXR", "AIF1 Capture", 1, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF2RXL", "AIF1 Playback", 2, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF2RXR", "AIF1 Playback", 3, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF2TXL", "AIF1 Capture", 2, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF2TXR", "AIF1 Capture", 3, SND_SOC_NOPM, 0, 0),
+
+	/* Output Side */
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC 0", NULL, RT274_SET_STREAMID_DAC0, 4, 0),
+	SND_SOC_DAPM_DAC("DAC 1", NULL, RT274_SET_STREAMID_DAC1, 4, 0),
+
+	/* Output Mux */
+	SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt274_hpo_mux),
+	SND_SOC_DAPM_MUX("LOUT Mux", SND_SOC_NOPM, 0, 0, &rt274_lout_mux),
+
+	SND_SOC_DAPM_SUPPLY("HP Power", RT274_SET_PIN_HPO,
+		RT274_SET_PIN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("LOUT Power", RT274_SET_PIN_LOUT3,
+		RT274_SET_PIN_SFT, 0, NULL, 0),
+
+	/* Output Mixer */
+	SND_SOC_DAPM_PGA("DAC OUT0", SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_PGA("DAC OUT1", SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	/* Output Pga */
+	SND_SOC_DAPM_SWITCH("LOUT L", SND_SOC_NOPM, 0, 0,
+		&loutl_enable_control),
+	SND_SOC_DAPM_SWITCH("LOUT R", SND_SOC_NOPM, 0, 0,
+		&loutr_enable_control),
+	SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0,
+		&hpol_enable_control),
+	SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0,
+		&hpor_enable_control),
+
+	/* Output Lines */
+	SND_SOC_DAPM_OUTPUT("HPO Pin"),
+	SND_SOC_DAPM_OUTPUT("SPDIF"),
+	SND_SOC_DAPM_OUTPUT("LINE3"),
+};
+
+static const struct snd_soc_dapm_route rt274_dapm_routes[] = {
+	{"DMIC1", NULL, "DMIC1 Pin"},
+	{"DMIC2", NULL, "DMIC2 Pin"},
+
+	{"ADC 0 Mux", "Mic", "MIC"},
+	{"ADC 0 Mux", "Dmic", "DMIC1"},
+	{"ADC 0 Mux", "Line1", "LINE1"},
+	{"ADC 0 Mux", "Line2", "LINE2"},
+	{"ADC 1 Mux", "Mic", "MIC"},
+	{"ADC 1 Mux", "Dmic", "DMIC2"},
+	{"ADC 1 Mux", "Line1", "LINE1"},
+	{"ADC 1 Mux", "Line2", "LINE2"},
+
+	{"ADC 0", NULL, "ADC 0 Mux"},
+	{"ADC 1", NULL, "ADC 1 Mux"},
+
+	{"AIF1TXL", NULL, "ADC 0"},
+	{"AIF1TXR", NULL, "ADC 0"},
+	{"AIF2TXL", NULL, "ADC 1"},
+	{"AIF2TXR", NULL, "ADC 1"},
+
+	{"DAC 0", NULL, "AIF1RXL"},
+	{"DAC 0", NULL, "AIF1RXR"},
+	{"DAC 1", NULL, "AIF2RXL"},
+	{"DAC 1", NULL, "AIF2RXR"},
+
+	{"DAC OUT0", NULL, "DAC 0"},
+
+	{"DAC OUT1", NULL, "DAC 1"},
+
+	{"LOUT Mux", "DAC OUT0", "DAC OUT0"},
+	{"LOUT Mux", "DAC OUT1", "DAC OUT1"},
+
+	{"LOUT L", "Switch", "LOUT Mux"},
+	{"LOUT R", "Switch", "LOUT Mux"},
+	{"LOUT L", NULL, "LOUT Power"},
+	{"LOUT R", NULL, "LOUT Power"},
+
+	{"LINE3", NULL, "LOUT L"},
+	{"LINE3", NULL, "LOUT R"},
+
+	{"HPO Mux", "DAC OUT0", "DAC OUT0"},
+	{"HPO Mux", "DAC OUT1", "DAC OUT1"},
+
+	{"HPO L", "Switch", "HPO Mux"},
+	{"HPO R", "Switch", "HPO Mux"},
+	{"HPO L", NULL, "HP Power"},
+	{"HPO R", NULL, "HP Power"},
+
+	{"HPO Pin", NULL, "HPO L"},
+	{"HPO Pin", NULL, "HPO R"},
+};
+
+static int rt274_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val = 0;
+	int d_len_code = 0, c_len_code = 0;
+
+	switch (params_rate(params)) {
+	/* bit 14 0:48K 1:44.1K */
+	case 44100:
+	case 48000:
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported sample rate %d\n",
+					params_rate(params));
+		return -EINVAL;
+	}
+	switch (rt274->sys_clk) {
+	case 12288000:
+	case 24576000:
+		if (params_rate(params) != 48000) {
+			dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+					params_rate(params), rt274->sys_clk);
+			return -EINVAL;
+		}
+		break;
+	case 11289600:
+	case 22579200:
+		if (params_rate(params) != 44100) {
+			dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+					params_rate(params), rt274->sys_clk);
+			return -EINVAL;
+		}
+		break;
+	}
+
+	if (params_channels(params) <= 16) {
+		/* bit 3:0 Number of Channel */
+		val |= (params_channels(params) - 1);
+	} else {
+		dev_err(codec->dev, "Unsupported channels %d\n",
+					params_channels(params));
+		return -EINVAL;
+	}
+
+	switch (params_width(params)) {
+	/* bit 6:4 Bits per Sample */
+	case 16:
+		d_len_code = 0;
+		c_len_code = 0;
+		val |= (0x1 << 4);
+		break;
+	case 32:
+		d_len_code = 2;
+		c_len_code = 3;
+		val |= (0x4 << 4);
+		break;
+	case 20:
+		d_len_code = 1;
+		c_len_code = 1;
+		val |= (0x2 << 4);
+		break;
+	case 24:
+		d_len_code = 2;
+		c_len_code = 2;
+		val |= (0x3 << 4);
+		break;
+	case 8:
+		d_len_code = 3;
+		c_len_code = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (rt274->master)
+		c_len_code = 0x3;
+
+	snd_soc_update_bits(codec,
+		RT274_I2S_CTRL1, 0xc018, d_len_code << 3 | c_len_code << 14);
+	dev_dbg(codec->dev, "format val = 0x%x\n", val);
+
+	snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x407f, val);
+	snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x407f, val);
+
+	return 0;
+}
+
+static int rt274_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_M);
+		rt274->master = true;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_S);
+		rt274->master = false;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		snd_soc_update_bits(codec, RT274_I2S_CTRL1,
+					RT274_I2S_FMT_MASK, RT274_I2S_FMT_I2S);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		snd_soc_update_bits(codec, RT274_I2S_CTRL1,
+					RT274_I2S_FMT_MASK, RT274_I2S_FMT_LJ);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		snd_soc_update_bits(codec, RT274_I2S_CTRL1,
+					RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMA);
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		snd_soc_update_bits(codec, RT274_I2S_CTRL1,
+					RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMB);
+		break;
+	default:
+		return -EINVAL;
+	}
+	/* bit 15 Stream Type 0:PCM 1:Non-PCM */
+	snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x8000, 0);
+	snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x8000, 0);
+
+	return 0;
+}
+
+static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
+			unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	switch (source) {
+	case RT274_PLL2_S_MCLK:
+		snd_soc_update_bits(codec, RT274_PLL2_CTRL,
+				RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_MCLK);
+		break;
+	default:
+		dev_warn(codec->dev, "invalid pll source, use BCLK\n");
+	case RT274_PLL2_S_BCLK:
+		snd_soc_update_bits(codec, RT274_PLL2_CTRL,
+				RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK);
+		break;
+	}
+
+	if (source == RT274_PLL2_S_BCLK) {
+		snd_soc_update_bits(codec, RT274_MCLK_CTRL,
+				(0x3 << 12), (0x3 << 12));
+		switch (rt274->fs) {
+		case 50:
+			snd_soc_write(codec, 0x7a, 0xaab6);
+			snd_soc_write(codec, 0x7b, 0x0301);
+			snd_soc_write(codec, 0x7c, 0x04fe);
+			break;
+		case 64:
+			snd_soc_write(codec, 0x7a, 0xaa96);
+			snd_soc_write(codec, 0x7b, 0x8003);
+			snd_soc_write(codec, 0x7c, 0x081e);
+			break;
+		case 128:
+			snd_soc_write(codec, 0x7a, 0xaa96);
+			snd_soc_write(codec, 0x7b, 0x8003);
+			snd_soc_write(codec, 0x7c, 0x080e);
+			break;
+		default:
+			dev_warn(codec->dev, "invalid freq_in, assume 4.8M\n");
+		case 100:
+			snd_soc_write(codec, 0x7a, 0xaab6);
+			snd_soc_write(codec, 0x7b, 0x0301);
+			snd_soc_write(codec, 0x7c, 0x047e);
+			break;
+		}
+	}
+
+	return 0;
+}
+
+static int rt274_set_dai_sysclk(struct snd_soc_dai *dai,
+				int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+	unsigned int clk_src, mclk_en;
+
+	dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq);
+
+	switch (clk_id) {
+	case RT274_SCLK_S_MCLK:
+		mclk_en = RT274_MCLK_MODE_EN;
+		clk_src = RT274_CLK_SRC_MCLK;
+		break;
+	case RT274_SCLK_S_PLL1:
+		mclk_en = RT274_MCLK_MODE_DIS;
+		clk_src = RT274_CLK_SRC_MCLK;
+		break;
+	case RT274_SCLK_S_PLL2:
+		mclk_en = RT274_MCLK_MODE_EN;
+		clk_src = RT274_CLK_SRC_PLL2;
+		break;
+	default:
+		mclk_en = RT274_MCLK_MODE_DIS;
+		clk_src = RT274_CLK_SRC_MCLK;
+		dev_warn(codec->dev, "invalid sysclk source, use PLL1\n");
+		break;
+	}
+	snd_soc_update_bits(codec, RT274_MCLK_CTRL,
+			RT274_MCLK_MODE_MASK, mclk_en);
+	snd_soc_update_bits(codec, RT274_CLK_CTRL,
+			RT274_CLK_SRC_MASK, clk_src);
+
+	switch (freq) {
+	case 19200000:
+		if (clk_id == RT274_SCLK_S_MCLK) {
+			dev_err(codec->dev, "Should not use MCLK\n");
+			return -EINVAL;
+		}
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL2, 0x40, 0x40);
+		break;
+	case 24000000:
+		if (clk_id == RT274_SCLK_S_MCLK) {
+			dev_err(codec->dev, "Should not use MCLK\n");
+			return -EINVAL;
+		}
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL2, 0x40, 0x0);
+		break;
+	case 12288000:
+	case 11289600:
+		snd_soc_update_bits(codec,
+			RT274_MCLK_CTRL, 0x1fcf, 0x0008);
+		break;
+	case 24576000:
+	case 22579200:
+		snd_soc_update_bits(codec,
+			RT274_MCLK_CTRL, 0x1fcf, 0x1543);
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported system clock\n");
+		return -EINVAL;
+	}
+
+	rt274->sys_clk = freq;
+	rt274->clk_id = clk_id;
+
+	return 0;
+}
+
+static int rt274_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio);
+	rt274->fs = ratio;
+	if ((ratio / 50) == 0)
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, 0x1000, 0x1000);
+	else
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, 0x1000, 0x0);
+
+
+	return 0;
+}
+
+static int rt274_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+			unsigned int rx_mask, int slots, int slot_width)
+
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	if (rx_mask || tx_mask) {
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_EN);
+	} else {
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_DIS);
+		return 0;
+	}
+
+	switch (slots) {
+	case 4:
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_4CH);
+		break;
+	case 2:
+		snd_soc_update_bits(codec,
+			RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_2CH);
+		break;
+	default:
+		dev_err(codec->dev,
+			"Support 2 or 4 slots TDM only\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int rt274_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (SND_SOC_BIAS_STANDBY ==
+			snd_soc_codec_get_bias_level(codec)) {
+			snd_soc_write(codec,
+				RT274_SET_AUDIO_POWER, AC_PWRST_D0);
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_write(codec,
+			RT274_SET_AUDIO_POWER, AC_PWRST_D3);
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static irqreturn_t rt274_irq(int irq, void *data)
+{
+	struct rt274_priv *rt274 = data;
+	bool hp = false;
+	bool mic = false;
+	int ret, status = 0;
+
+	/* Clear IRQ */
+	regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
+				RT274_IRQ_CLR, RT274_IRQ_CLR);
+
+	ret = rt274_jack_detect(rt274, &hp, &mic);
+
+	if (ret == 0) {
+		if (hp == true)
+			status |= SND_JACK_HEADPHONE;
+
+		if (mic == true)
+			status |= SND_JACK_MICROPHONE;
+
+		snd_soc_jack_report(rt274->jack, status,
+			SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+
+		pm_wakeup_event(&rt274->i2c->dev, 300);
+	}
+
+	return IRQ_HANDLED;
+}
+
+static int rt274_probe(struct snd_soc_codec *codec)
+{
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	rt274->codec = codec;
+
+	if (rt274->i2c->irq) {
+		INIT_DELAYED_WORK(&rt274->jack_detect_work,
+					rt274_jack_detect_work);
+		schedule_delayed_work(&rt274->jack_detect_work,
+					msecs_to_jiffies(1250));
+	}
+
+	return 0;
+}
+
+static int rt274_remove(struct snd_soc_codec *codec)
+{
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	cancel_delayed_work_sync(&rt274->jack_detect_work);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int rt274_suspend(struct snd_soc_codec *codec)
+{
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	regcache_cache_only(rt274->regmap, true);
+	regcache_mark_dirty(rt274->regmap);
+
+	return 0;
+}
+
+static int rt274_resume(struct snd_soc_codec *codec)
+{
+	struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+
+	regcache_cache_only(rt274->regmap, false);
+	rt274_index_sync(codec);
+	regcache_sync(rt274->regmap);
+
+	return 0;
+}
+#else
+#define rt274_suspend NULL
+#define rt274_resume NULL
+#endif
+
+#define RT274_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+#define RT274_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static const struct snd_soc_dai_ops rt274_aif_dai_ops = {
+	.hw_params = rt274_hw_params,
+	.set_fmt = rt274_set_dai_fmt,
+	.set_sysclk = rt274_set_dai_sysclk,
+	.set_pll = rt274_set_dai_pll,
+	.set_bclk_ratio = rt274_set_bclk_ratio,
+	.set_tdm_slot = rt274_set_tdm_slot,
+};
+
+static struct snd_soc_dai_driver rt274_dai[] = {
+	{
+		.name = "rt274-aif1",
+		.id = RT274_AIF1,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT274_STEREO_RATES,
+			.formats = RT274_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF1 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT274_STEREO_RATES,
+			.formats = RT274_FORMATS,
+		},
+		.ops = &rt274_aif_dai_ops,
+		.symmetric_rates = 1,
+	},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_rt274 = {
+	.probe = rt274_probe,
+	.remove = rt274_remove,
+	.suspend = rt274_suspend,
+	.resume = rt274_resume,
+	.set_bias_level = rt274_set_bias_level,
+	.idle_bias_off = true,
+	.component_driver = {
+		.controls		= rt274_snd_controls,
+		.num_controls		= ARRAY_SIZE(rt274_snd_controls),
+		.dapm_widgets		= rt274_dapm_widgets,
+		.num_dapm_widgets	= ARRAY_SIZE(rt274_dapm_widgets),
+		.dapm_routes		= rt274_dapm_routes,
+		.num_dapm_routes	= ARRAY_SIZE(rt274_dapm_routes),
+	},
+	.set_jack = rt274_mic_detect,
+};
+
+static const struct regmap_config rt274_regmap = {
+	.reg_bits = 32,
+	.val_bits = 32,
+	.max_register = 0x05bfffff,
+	.volatile_reg = rt274_volatile_register,
+	.readable_reg = rt274_readable_register,
+	.reg_write = rl6347a_hw_write,
+	.reg_read = rl6347a_hw_read,
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = rt274_reg,
+	.num_reg_defaults = ARRAY_SIZE(rt274_reg),
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id rt274_of_match[] = {
+	{.compatible = "realtek,rt274"},
+	{},
+};
+MODULE_DEVICE_TABLE(of, rt274_of_match);
+#endif
+
+static const struct i2c_device_id rt274_i2c_id[] = {
+	{"rt274", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, rt274_i2c_id);
+
+static const struct acpi_device_id rt274_acpi_match[] = {
+	{ "10EC0274", 0 },
+	{},
+};
+MODULE_DEVICE_TABLE(acpi, rt274_acpi_match);
+
+static int rt274_i2c_probe(struct i2c_client *i2c,
+			   const struct i2c_device_id *id)
+{
+	struct rt274_priv *rt274;
+
+	int ret;
+	unsigned int val;
+
+	rt274 = devm_kzalloc(&i2c->dev,	sizeof(*rt274),
+				GFP_KERNEL);
+	if (rt274 == NULL)
+		return -ENOMEM;
+
+	rt274->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt274_regmap);
+	if (IS_ERR(rt274->regmap)) {
+		ret = PTR_ERR(rt274->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+
+	regmap_read(rt274->regmap,
+		RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val);
+	if (val != RT274_VENDOR_ID) {
+		dev_err(&i2c->dev,
+			"Device with ID register %#x is not rt274\n", val);
+		return -ENODEV;
+	}
+
+	rt274->index_cache = devm_kmemdup(&i2c->dev, rt274_index_def,
+					  sizeof(rt274_index_def), GFP_KERNEL);
+	if (!rt274->index_cache)
+		return -ENOMEM;
+
+	rt274->index_cache_size = INDEX_CACHE_SIZE;
+	rt274->i2c = i2c;
+	i2c_set_clientdata(i2c, rt274);
+
+	/* reset codec */
+	regmap_write(rt274->regmap, RT274_RESET, 0);
+	regmap_update_bits(rt274->regmap, 0x1a, 0x4000, 0x4000);
+
+	/* Set Pad PDB is floating */
+	regmap_update_bits(rt274->regmap, RT274_PAD_CTRL12, 0x3, 0x0);
+	regmap_write(rt274->regmap, RT274_COEF5b_INDEX, 0x01);
+	regmap_write(rt274->regmap, RT274_COEF5b_COEF, 0x8540);
+	regmap_update_bits(rt274->regmap, 0x6f, 0x0100, 0x0100);
+	/* Combo jack auto detect */
+	regmap_write(rt274->regmap, 0x4a, 0x201b);
+	/* Aux mode off */
+	regmap_update_bits(rt274->regmap, 0x6f, 0x3000, 0x2000);
+	/* HP DC Calibration */
+	regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0x0);
+	//Set NID=58h.Index 00h [15]= 1b;
+	regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00);
+	regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888);
+	msleep(500);
+	regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0xb);
+	regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00);
+	regmap_write(rt274->regmap, RT274_COEF58_COEF, 0x3888);
+	/* Set pin widget */
+	regmap_write(rt274->regmap, RT274_SET_PIN_HPO, 0x40);
+	regmap_write(rt274->regmap, RT274_SET_PIN_LOUT3, 0x40);
+	regmap_write(rt274->regmap, RT274_SET_MIC, 0x20);
+	regmap_write(rt274->regmap, RT274_SET_PIN_DMIC1, 0x20);
+
+	regmap_update_bits(rt274->regmap, RT274_I2S_CTRL2, 0xc004, 0x4004);
+	regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
+				RT274_GPI2_SEL_MASK, RT274_GPI2_SEL_DMIC_CLK);
+
+	/* jack detection */
+	regmap_write(rt274->regmap, RT274_UNSOLICITED_HP_OUT, 0x81);
+	regmap_write(rt274->regmap, RT274_UNSOLICITED_MIC, 0x82);
+
+	if (rt274->i2c->irq) {
+		ret = request_threaded_irq(rt274->i2c->irq, NULL, rt274_irq,
+			IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt274", rt274);
+		if (ret != 0) {
+			dev_err(&i2c->dev,
+				"Failed to reguest IRQ: %d\n", ret);
+			return ret;
+		}
+	}
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt274,
+				     rt274_dai, ARRAY_SIZE(rt274_dai));
+
+	return ret;
+}
+
+static int rt274_i2c_remove(struct i2c_client *i2c)
+{
+	struct rt274_priv *rt274 = i2c_get_clientdata(i2c);
+
+	if (i2c->irq)
+		free_irq(i2c->irq, rt274);
+	snd_soc_unregister_codec(&i2c->dev);
+
+	return 0;
+}
+
+
+static struct i2c_driver rt274_i2c_driver = {
+	.driver = {
+		   .name = "rt274",
+		   .acpi_match_table = ACPI_PTR(rt274_acpi_match),
+#ifdef CONFIG_OF
+		   .of_match_table = of_match_ptr(rt274_of_match),
+#endif
+		   },
+	.probe = rt274_i2c_probe,
+	.remove = rt274_i2c_remove,
+	.id_table = rt274_i2c_id,
+};
+
+module_i2c_driver(rt274_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT274 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt274.h b/sound/soc/codecs/rt274.h
new file mode 100644
index 000000000000..4fd1bcb73dba
--- /dev/null
+++ b/sound/soc/codecs/rt274.h
@@ -0,0 +1,217 @@
+/*
+ * rt274.h  --  RT274 ALSA SoC audio driver
+ *
+ * Copyright 2016 Realtek Microelectronics
+ * Author: Bard Liao <bardliao@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __RT274_H__
+#define __RT274_H__
+
+#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
+
+#define RT274_AUDIO_FUNCTION_GROUP			0x01
+#define RT274_DAC_OUT0					0x02
+#define RT274_DAC_OUT1					0x03
+#define RT274_ADC_IN2					0x08
+#define RT274_ADC_IN1					0x09
+#define RT274_DIG_CVT					0x0a
+#define RT274_DMIC1					0x12
+#define RT274_DMIC2					0x13
+#define RT274_MIC					0x19
+#define RT274_LINE1					0x1a
+#define RT274_LINE2					0x1b
+#define RT274_LINE3					0x16
+#define RT274_SPDIF					0x1e
+#define RT274_VENDOR_REGISTERS				0x20
+#define RT274_HP_OUT					0x21
+#define RT274_MIXER_IN1					0x22
+#define RT274_MIXER_IN2					0x23
+#define RT274_INLINE_CMD				0x55
+
+#define RT274_SET_PIN_SFT				6
+#define RT274_SET_PIN_ENABLE				0x40
+#define RT274_SET_PIN_DISABLE				0
+#define RT274_SET_EAPD_HIGH				0x2
+#define RT274_SET_EAPD_LOW				0
+
+#define RT274_MUTE_SFT					7
+
+/* Verb commands */
+#define RT274_RESET\
+	VERB_CMD(AC_VERB_SET_CODEC_RESET, RT274_AUDIO_FUNCTION_GROUP, 0)
+#define RT274_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM)
+#define RT274_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0)
+#define RT274_SET_AUDIO_POWER RT274_SET_POWER(RT274_AUDIO_FUNCTION_GROUP)
+#define RT274_SET_HPO_POWER RT274_SET_POWER(RT274_HP_OUT)
+#define RT274_SET_DMIC1_POWER RT274_SET_POWER(RT274_DMIC1)
+#define RT274_LOUT_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_LINE3, 0)
+#define RT274_HPO_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_HP_OUT, 0)
+#define RT274_ADC0_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN1, 0)
+#define RT274_ADC1_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN2, 0)
+#define RT274_SET_MIC\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_MIC, 0)
+#define RT274_SET_PIN_LOUT3\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_LINE3, 0)
+#define RT274_SET_PIN_HPO\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0)
+#define RT274_SET_PIN_DMIC1\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0)
+#define RT274_SET_PIN_SPDIF\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_SPDIF, 0)
+#define RT274_SET_PIN_DIG_CVT\
+	VERB_CMD(AC_VERB_SET_DIGI_CONVERT_1, RT274_DIG_CVT, 0)
+#define RT274_SET_AMP_GAIN_HPO\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0)
+#define RT274_SET_AMP_GAIN_ADC_IN1\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0)
+#define RT274_SET_AMP_GAIN_ADC_IN2\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0)
+#define RT274_GET_HP_SENSE\
+	VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_HP_OUT, 0)
+#define RT274_GET_MIC_SENSE\
+	VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_MIC, 0)
+#define RT274_SET_DMIC2_DEFAULT\
+	VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_DMIC2, 0)
+#define RT274_SET_SPDIF_DEFAULT\
+	VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_SPDIF, 0)
+#define RT274_DAC0L_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0xa000)
+#define RT274_DAC0R_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0x9000)
+#define RT274_DAC1L_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0xa000)
+#define RT274_DAC1R_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0x9000)
+#define RT274_ADCL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x6000)
+#define RT274_ADCR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x5000)
+#define RT274_MIC_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_MIC, 0x7000)
+#define RT274_LOUTL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0xa000)
+#define RT274_LOUTR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0x9000)
+#define RT274_HPOL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0xa000)
+#define RT274_HPOR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0x9000)
+#define RT274_DAC_FORMAT\
+	VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_DAC_OUT0, 0)
+#define RT274_ADC_FORMAT\
+	VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_ADC_IN1, 0)
+#define RT274_COEF_INDEX\
+	VERB_CMD(AC_VERB_SET_COEF_INDEX, RT274_VENDOR_REGISTERS, 0)
+#define RT274_PROC_COEF\
+	VERB_CMD(AC_VERB_SET_PROC_COEF, RT274_VENDOR_REGISTERS, 0)
+#define RT274_UNSOLICITED_INLINE_CMD\
+	VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_INLINE_CMD, 0)
+#define RT274_UNSOLICITED_HP_OUT\
+	VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_HP_OUT, 0)
+#define RT274_UNSOLICITED_MIC\
+	VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_MIC, 0)
+#define RT274_COEF58_INDEX\
+	VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x58, 0)
+#define RT274_COEF58_COEF\
+	VERB_CMD(AC_VERB_SET_PROC_COEF, 0x58, 0)
+#define RT274_COEF5b_INDEX\
+	VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x5b, 0)
+#define RT274_COEF5b_COEF\
+	VERB_CMD(AC_VERB_SET_PROC_COEF, 0x5b, 0)
+#define RT274_SET_STREAMID_DAC0\
+	VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT0, 0)
+#define RT274_SET_STREAMID_DAC1\
+	VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT1, 0)
+#define RT274_SET_STREAMID_ADC1\
+	VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN1, 0)
+#define RT274_SET_STREAMID_ADC2\
+	VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN2, 0)
+
+/* Index registers */
+#define RT274_EAPD_GPIO_IRQ_CTRL	0x10
+#define RT274_PAD_CTRL12		0x35
+#define RT274_I2S_CTRL1			0x63
+#define RT274_I2S_CTRL2			0x64
+#define RT274_MCLK_CTRL			0x71
+#define RT274_CLK_CTRL			0x72
+#define RT274_PLL2_CTRL			0x7b
+
+
+/* EAPD GPIO IRQ control (Index 0x10) */
+#define RT274_IRQ_DIS		(0x0 << 13)
+#define RT274_IRQ_EN		(0x1 << 13)
+#define RT274_IRQ_CLR		(0x1 << 12)
+#define RT274_GPI2_SEL_MASK	(0x3 << 7)
+#define RT274_GPI2_SEL_GPIO2	(0x0 << 7)
+#define RT274_GPI2_SEL_I2S	(0x1 << 7)
+#define RT274_GPI2_SEL_DMIC_CLK	(0x2 << 7)
+#define RT274_GPI2_SEL_CBJ	(0x3 << 7)
+
+/* Front I2S_Interface control 1 (Index 0x63) */
+#define RT274_I2S_MODE_MASK	(0x1 << 11)
+#define RT274_I2S_MODE_S	(0x0 << 11)
+#define RT274_I2S_MODE_M	(0x1 << 11)
+#define RT274_TDM_DIS		(0x0 << 10)
+#define RT274_TDM_EN		(0x1 << 10)
+#define RT274_TDM_CH_NUM	(0x1 << 7)
+#define RT274_TDM_2CH		(0x0 << 7)
+#define RT274_TDM_4CH		(0x1 << 7)
+#define RT274_I2S_FMT_MASK	(0x3 << 8)
+#define RT274_I2S_FMT_I2S	(0x0 << 8)
+#define RT274_I2S_FMT_LJ	(0x1 << 8)
+#define RT274_I2S_FMT_PCMA	(0x2 << 8)
+#define RT274_I2S_FMT_PCMB	(0x3 << 8)
+
+/* MCLK clock domain control (Index 0x71) */
+#define RT274_MCLK_MODE_MASK	(0x1 << 14)
+#define RT274_MCLK_MODE_DIS	(0x0 << 14)
+#define RT274_MCLK_MODE_EN	(0x1 << 14)
+
+/* Clock control (Index 0x72) */
+#define RT274_CLK_SRC_MASK	(0x7 << 3)
+#define RT274_CLK_SRC_MCLK	(0x0 << 3)
+#define RT274_CLK_SRC_PLL2	(0x3 << 3)
+
+/* PLL2 control (Index 0x7b) */
+#define RT274_PLL2_SRC_MASK	(0x1 << 13)
+#define RT274_PLL2_SRC_MCLK	(0x0 << 13)
+#define RT274_PLL2_SRC_BCLK	(0x1 << 13)
+
+/* HP-OUT (0x21) */
+#define RT274_M_HP_MUX_SFT	14
+#define RT274_HP_SEL_MASK	0x1
+#define RT274_HP_SEL_SFT	0
+#define RT274_HP_SEL_F		0
+#define RT274_HP_SEL_S		1
+
+/* ADC (0x22) (0x23) */
+#define RT274_ADC_SEL_MASK	0x7
+#define RT274_ADC_SEL_SFT	0
+#define RT274_ADC_SEL_MIC	0
+#define RT274_ADC_SEL_LINE1	1
+#define RT274_ADC_SEL_LINE2	2
+#define RT274_ADC_SEL_DMIC	3
+
+#define RT274_SCLK_S_MCLK	0
+#define RT274_SCLK_S_PLL1	1
+#define RT274_SCLK_S_PLL2	2
+
+#define RT274_PLL2_S_MCLK	0
+#define RT274_PLL2_S_BCLK	1
+
+enum {
+	RT274_AIF1,
+	RT274_AIFS,
+};
+
+#endif /* __RT274_H__ */
+
-- 
cgit v1.2.3


From 49a69163ddbe47063be960ad5c2fcfe94c8b473b Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Wed, 19 Jul 2017 09:52:12 +0800
Subject: ASoC: rt274: correct comment style

There was a comment style issue in the driver.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt274.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index 81e29ba2f50c..fb683ffc4e03 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -1159,7 +1159,7 @@ static int rt274_i2c_probe(struct i2c_client *i2c,
 	regmap_update_bits(rt274->regmap, 0x6f, 0x3000, 0x2000);
 	/* HP DC Calibration */
 	regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0x0);
-	//Set NID=58h.Index 00h [15]= 1b;
+	/* Set NID=58h.Index 00h [15]= 1b; */
 	regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00);
 	regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888);
 	msleep(500);
-- 
cgit v1.2.3


From 1d5c5b6582cf829fc4c0c8eef6b9f9f0dc29c7b2 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Wed, 19 Jul 2017 10:20:23 +0800
Subject: ASoC: rt5663: Correct the mixer switch setting and remove redundant
 routing path

The patch corrects the mixer siwtch setting that was the mistake in the
previous commitment. And remove the redundant routing paths.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index d3f7bc2173e8..45f8a9888f7d 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -59,7 +59,7 @@ struct rt5663_priv {
 };
 
 static const struct reg_sequence rt5663_patch_list[] = {
-	{ 0x002a, 0x8080 },
+	{ 0x002a, 0x8020 },
 	{ 0x0086, 0x0028 },
 };
 
@@ -482,7 +482,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0023, 0x0039 },
 	{ 0x0026, 0xc0c0 },
 	{ 0x0029, 0x8080 },
-	{ 0x002a, 0x8080 },
+	{ 0x002a, 0x8020 },
 	{ 0x002c, 0x000c },
 	{ 0x002d, 0x0000 },
 	{ 0x0040, 0x0808 },
@@ -1967,8 +1967,8 @@ static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = {
 };
 
 static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = {
-	SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER,
-			RT5663_M_DAC_L1_STO_R_SHIFT, 1, 1),
+	SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER,
+			RT5663_M_DAC_R1_STO_R_SHIFT, 1, 1),
 };
 
 /* Out Switch */
@@ -2358,11 +2358,9 @@ static const struct snd_soc_dapm_route rt5663_dapm_routes[] = {
 	{ "DAC R1", NULL, "ADDA MIXR" },
 
 	{ "STO1 DAC MIXL", "DAC L Switch", "DAC L1" },
-	{ "STO1 DAC MIXL", "DAC R Switch", "DAC R1" },
 	{ "STO1 DAC MIXL", NULL, "STO1 DAC L Power" },
 	{ "STO1 DAC MIXL", NULL, "STO1 DAC Filter" },
 	{ "STO1 DAC MIXR", "DAC R Switch", "DAC R1" },
-	{ "STO1 DAC MIXR", "DAC L Switch", "DAC L1" },
 	{ "STO1 DAC MIXR", NULL, "STO1 DAC R Power" },
 	{ "STO1 DAC MIXR", NULL, "STO1 DAC Filter" },
 
-- 
cgit v1.2.3


From ddc9e69b9dc23d4c0d8ed829575327746ea77a04 Mon Sep 17 00:00:00 2001
From: Andy Shevchenko
Date: Tue, 18 Jul 2017 20:34:17 +0300
Subject: ASoC: rt5677: Hide platform data in the module sources

There is no user of legacy platform data.

Remove separate header and hide its content inside module sources.

Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/rt5677.h    | 45 ---------------------------------------------
 sound/soc/codecs/rt5677.c | 29 +++++++++++++++++------------
 sound/soc/codecs/rt5677.h | 30 +++++++++++++++++++++++++++++-
 3 files changed, 46 insertions(+), 58 deletions(-)
 delete mode 100644 include/sound/rt5677.h

(limited to 'sound')

diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h
deleted file mode 100644
index a6207043ac3c..000000000000
--- a/include/sound/rt5677.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * linux/sound/rt5677.h -- Platform data for RT5677
- *
- * Copyright 2013 Realtek Semiconductor Corp.
- * Author: Oder Chiou <oder_chiou@realtek.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef __LINUX_SND_RT5677_H
-#define __LINUX_SND_RT5677_H
-
-enum rt5677_dmic2_clk {
-	RT5677_DMIC_CLK1 = 0,
-	RT5677_DMIC_CLK2 = 1,
-};
-
-
-struct rt5677_platform_data {
-	/* IN1/IN2/LOUT1/LOUT2/LOUT3 can optionally be differential */
-	bool in1_diff;
-	bool in2_diff;
-	bool lout1_diff;
-	bool lout2_diff;
-	bool lout3_diff;
-	/* DMIC2 clock source selection */
-	enum rt5677_dmic2_clk dmic2_clk_pin;
-
-	/* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */
-	u8 gpio_config[6];
-
-	/* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */
-	unsigned int jd1_gpio;
-	/* jd2 and jd3 can select 0 ~ 3 as
-		OFF, GPIO4, GPIO5 and GPIO6 respectively */
-	unsigned int jd2_gpio;
-	unsigned int jd3_gpio;
-
-	/* Set MICBIAS1 VDD 1v8 or 3v3 */
-	bool micbias1_vdd_3v3;
-};
-
-#endif
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 36e530a36c82..0f9642c2198a 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -21,6 +21,7 @@
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/firmware.h>
+#include <linux/of_device.h>
 #include <linux/property.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -5019,25 +5020,21 @@ static const struct regmap_config rt5677_regmap = {
 };
 
 static const struct i2c_device_id rt5677_i2c_id[] = {
-	{ "rt5677", RT5677 },
-	{ "rt5676", RT5676 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id);
 
 static const struct of_device_id rt5677_of_match[] = {
-	{ .compatible = "realtek,rt5677", },
+	{ .compatible = "realtek,rt5677", RT5677 },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, rt5677_of_match);
 
-#ifdef CONFIG_ACPI
 static const struct acpi_device_id rt5677_acpi_match[] = {
 	{ "RT5677CE", RT5677 },
 	{ }
 };
 MODULE_DEVICE_TABLE(acpi, rt5677_acpi_match);
-#endif
 
 static void rt5677_read_acpi_properties(struct rt5677_priv *rt5677,
 		struct device *dev)
@@ -5147,7 +5144,6 @@ static void rt5677_free_irq(struct i2c_client *i2c)
 static int rt5677_i2c_probe(struct i2c_client *i2c,
 		    const struct i2c_device_id *id)
 {
-	struct rt5677_platform_data *pdata = dev_get_platdata(&i2c->dev);
 	struct rt5677_priv *rt5677;
 	int ret;
 	unsigned int val;
@@ -5159,16 +5155,25 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, rt5677);
 
-	rt5677->type = id->driver_data;
+	if (i2c->dev.of_node) {
+		const struct of_device_id *match_id;
+
+		match_id = of_match_device(rt5677_of_match, &i2c->dev);
+		if (match_id)
+			rt5677->type = (enum rt5677_type)match_id->data;
 
-	if (pdata)
-		rt5677->pdata = *pdata;
-	else if (i2c->dev.of_node)
 		rt5677_read_device_properties(rt5677, &i2c->dev);
-	else if (ACPI_HANDLE(&i2c->dev))
+	} else if (ACPI_HANDLE(&i2c->dev)) {
+		const struct acpi_device_id *acpi_id;
+
+		acpi_id = acpi_match_device(rt5677_acpi_match, &i2c->dev);
+		if (acpi_id)
+			rt5677->type = (enum rt5677_type)acpi_id->driver_data;
+
 		rt5677_read_acpi_properties(rt5677, &i2c->dev);
-	else
+	} else {
 		return -EINVAL;
+	}
 
 	/* pow-ldo2 and reset are optional. The codec pins may be statically
 	 * connected on the board without gpios. If the gpio device property
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index d46855a42c40..97239973edc4 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -12,7 +12,6 @@
 #ifndef __RT5677_H__
 #define __RT5677_H__
 
-#include <sound/rt5677.h>
 #include <linux/gpio/driver.h>
 #include <linux/gpio/consumer.h>
 
@@ -1761,6 +1760,35 @@ enum {
 	RT5677_I2S4_SOURCE = (0x1 << 18),
 };
 
+enum rt5677_dmic2_clk {
+	RT5677_DMIC_CLK1 = 0,
+	RT5677_DMIC_CLK2 = 1,
+};
+
+struct rt5677_platform_data {
+	/* IN1/IN2/LOUT1/LOUT2/LOUT3 can optionally be differential */
+	bool in1_diff;
+	bool in2_diff;
+	bool lout1_diff;
+	bool lout2_diff;
+	bool lout3_diff;
+	/* DMIC2 clock source selection */
+	enum rt5677_dmic2_clk dmic2_clk_pin;
+
+	/* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */
+	u8 gpio_config[6];
+
+	/* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */
+	unsigned int jd1_gpio;
+	/* jd2 and jd3 can select 0 ~ 3 as
+		OFF, GPIO4, GPIO5 and GPIO6 respectively */
+	unsigned int jd2_gpio;
+	unsigned int jd3_gpio;
+
+	/* Set MICBIAS1 VDD 1v8 or 3v3 */
+	bool micbias1_vdd_3v3;
+};
+
 struct rt5677_priv {
 	struct snd_soc_codec *codec;
 	struct rt5677_platform_data pdata;
-- 
cgit v1.2.3


From d4e753d3aa2c278cb1f7b4a61987c4037316e0fd Mon Sep 17 00:00:00 2001
From: Andy Shevchenko
Date: Tue, 18 Jul 2017 20:34:18 +0300
Subject: ASoC: rt5677: Refactor code to avoid comparison unsigned >= 0

rt5677_to_irq() has couple of redundant conditionals, one of which
compares unsigned variable to be great than or equal to zero which
is always true as compiler notices:

sound/soc/codecs/rt5677.c: In function ‘rt5677_to_irq’:
sound/soc/codecs/rt5677.c:4626:13: warning: comparison of unsigned expression >= 0 is always true [-Wtype-limits]

Refactor the code by removing redundant conditionals.

Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 50 ++++++++++++++++++++---------------------------
 1 file changed, 21 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 0f9642c2198a..a3cce57739c3 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4625,35 +4625,27 @@ static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset)
 	struct regmap_irq_chip_data *data = rt5677->irq_data;
 	int irq;
 
-	if (offset >= RT5677_GPIO1 && offset <= RT5677_GPIO3) {
-		if ((rt5677->pdata.jd1_gpio == 1 && offset == RT5677_GPIO1) ||
-			(rt5677->pdata.jd1_gpio == 2 &&
-				offset == RT5677_GPIO2) ||
-			(rt5677->pdata.jd1_gpio == 3 &&
-				offset == RT5677_GPIO3)) {
-			irq = RT5677_IRQ_JD1;
-		} else {
-			return -ENXIO;
-		}
-	}
-
-	if (offset >= RT5677_GPIO4 && offset <= RT5677_GPIO6) {
-		if ((rt5677->pdata.jd2_gpio == 1 && offset == RT5677_GPIO4) ||
-			(rt5677->pdata.jd2_gpio == 2 &&
-				offset == RT5677_GPIO5) ||
-			(rt5677->pdata.jd2_gpio == 3 &&
-				offset == RT5677_GPIO6)) {
-			irq = RT5677_IRQ_JD2;
-		} else if ((rt5677->pdata.jd3_gpio == 1 &&
-				offset == RT5677_GPIO4) ||
-			(rt5677->pdata.jd3_gpio == 2 &&
-				offset == RT5677_GPIO5) ||
-			(rt5677->pdata.jd3_gpio == 3 &&
-				offset == RT5677_GPIO6)) {
-			irq = RT5677_IRQ_JD3;
-		} else {
-			return -ENXIO;
-		}
+	if ((rt5677->pdata.jd1_gpio == 1 && offset == RT5677_GPIO1) ||
+		(rt5677->pdata.jd1_gpio == 2 &&
+			offset == RT5677_GPIO2) ||
+		(rt5677->pdata.jd1_gpio == 3 &&
+			offset == RT5677_GPIO3)) {
+		irq = RT5677_IRQ_JD1;
+	} else if ((rt5677->pdata.jd2_gpio == 1 && offset == RT5677_GPIO4) ||
+		(rt5677->pdata.jd2_gpio == 2 &&
+			offset == RT5677_GPIO5) ||
+		(rt5677->pdata.jd2_gpio == 3 &&
+			offset == RT5677_GPIO6)) {
+		irq = RT5677_IRQ_JD2;
+	} else if ((rt5677->pdata.jd3_gpio == 1 &&
+			offset == RT5677_GPIO4) ||
+		(rt5677->pdata.jd3_gpio == 2 &&
+			offset == RT5677_GPIO5) ||
+		(rt5677->pdata.jd3_gpio == 3 &&
+			offset == RT5677_GPIO6)) {
+		irq = RT5677_IRQ_JD3;
+	} else {
+		return -ENXIO;
 	}
 
 	return regmap_irq_get_virq(data, irq);
-- 
cgit v1.2.3


From 3a4e1b93110362f0ca6f81c564b04b2aa4bab853 Mon Sep 17 00:00:00 2001
From: Philipp Zabel
Date: Wed, 19 Jul 2017 17:26:41 +0200
Subject: ASoC: img: explicitly request exclusive reset control

Commit a53e35db70d1 ("reset: Ensure drivers are explicit when requesting
reset lines") started to transition the reset control request API calls
to explicitly state whether the driver needs exclusive or shared reset
control behavior. Convert all drivers requesting exclusive resets to the
explicit API call so the temporary transition helpers can be removed.

No functional changes.

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/img/img-i2s-in.c       | 2 +-
 sound/soc/img/img-i2s-out.c      | 2 +-
 sound/soc/img/img-parallel-out.c | 2 +-
 sound/soc/img/img-spdif-in.c     | 2 +-
 sound/soc/img/img-spdif-out.c    | 2 +-
 5 files changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c
index 0389203f8560..567f9767fb73 100644
--- a/sound/soc/img/img-i2s-in.c
+++ b/sound/soc/img/img-i2s-in.c
@@ -443,7 +443,7 @@ static int img_i2s_in_probe(struct platform_device *pdev)
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE;
 	i2s->dai_driver.ops = &img_i2s_in_dai_ops;
 
-	rst = devm_reset_control_get(dev, "rst");
+	rst = devm_reset_control_get_exclusive(dev, "rst");
 	if (IS_ERR(rst)) {
 		if (PTR_ERR(rst) == -EPROBE_DEFER) {
 			ret = -EPROBE_DEFER;
diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c
index 5f997135a8ae..78b7f6cd675b 100644
--- a/sound/soc/img/img-i2s-out.c
+++ b/sound/soc/img/img-i2s-out.c
@@ -446,7 +446,7 @@ static int img_i2s_out_probe(struct platform_device *pdev)
 
 	i2s->channel_base = base + (max_i2s_chan_pow_2 * 0x20);
 
-	i2s->rst = devm_reset_control_get(&pdev->dev, "rst");
+	i2s->rst = devm_reset_control_get_exclusive(&pdev->dev, "rst");
 	if (IS_ERR(i2s->rst)) {
 		if (PTR_ERR(i2s->rst) != -EPROBE_DEFER)
 			dev_err(&pdev->dev, "No top level reset found\n");
diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c
index 33ceb207ee70..23b0f0f6ec9c 100644
--- a/sound/soc/img/img-parallel-out.c
+++ b/sound/soc/img/img-parallel-out.c
@@ -224,7 +224,7 @@ static int img_prl_out_probe(struct platform_device *pdev)
 
 	prl->base = base;
 
-	prl->rst = devm_reset_control_get(&pdev->dev, "rst");
+	prl->rst = devm_reset_control_get_exclusive(&pdev->dev, "rst");
 	if (IS_ERR(prl->rst)) {
 		if (PTR_ERR(prl->rst) != -EPROBE_DEFER)
 			dev_err(&pdev->dev, "No top level reset found\n");
diff --git a/sound/soc/img/img-spdif-in.c b/sound/soc/img/img-spdif-in.c
index 4d9953d318af..8adfd65d4390 100644
--- a/sound/soc/img/img-spdif-in.c
+++ b/sound/soc/img/img-spdif-in.c
@@ -727,7 +727,7 @@ static int img_spdif_in_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	rst = devm_reset_control_get(&pdev->dev, "rst");
+	rst = devm_reset_control_get_exclusive(&pdev->dev, "rst");
 	if (IS_ERR(rst)) {
 		if (PTR_ERR(rst) == -EPROBE_DEFER) {
 			ret = -EPROBE_DEFER;
diff --git a/sound/soc/img/img-spdif-out.c b/sound/soc/img/img-spdif-out.c
index 08f93a5dadfe..383655da2e60 100644
--- a/sound/soc/img/img-spdif-out.c
+++ b/sound/soc/img/img-spdif-out.c
@@ -334,7 +334,7 @@ static int img_spdif_out_probe(struct platform_device *pdev)
 
 	spdif->base = base;
 
-	spdif->rst = devm_reset_control_get(&pdev->dev, "rst");
+	spdif->rst = devm_reset_control_get_exclusive(&pdev->dev, "rst");
 	if (IS_ERR(spdif->rst)) {
 		if (PTR_ERR(spdif->rst) != -EPROBE_DEFER)
 			dev_err(&pdev->dev, "No top level reset found\n");
-- 
cgit v1.2.3


From 635eac1e54d82c59f621a0f38a9bffae50f150c7 Mon Sep 17 00:00:00 2001
From: Philipp Zabel
Date: Wed, 19 Jul 2017 17:26:42 +0200
Subject: ASoC: stm32: explicitly request exclusive reset control

Commit a53e35db70d1 ("reset: Ensure drivers are explicit when requesting
reset lines") started to transition the reset control request API calls
to explicitly state whether the driver needs exclusive or shared reset
control behavior. Convert all drivers requesting exclusive resets to the
explicit API call so the temporary transition helpers can be removed.

No functional changes.

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Maxime Coquelin <mcoquelin.stm32@gmail.com>
Cc: Alexandre Torgue <alexandre.torgue@st.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/stm/stm32_i2s.c     | 2 +-
 sound/soc/stm/stm32_sai.c     | 2 +-
 sound/soc/stm/stm32_spdifrx.c | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c
index 8052629a89df..6d0bf78d114d 100644
--- a/sound/soc/stm/stm32_i2s.c
+++ b/sound/soc/stm/stm32_i2s.c
@@ -840,7 +840,7 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev,
 	}
 
 	/* Reset */
-	rst = devm_reset_control_get(&pdev->dev, NULL);
+	rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
 	if (!IS_ERR(rst)) {
 		reset_control_assert(rst);
 		udelay(2);
diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index f7713314913b..1258bef4dcb3 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -85,7 +85,7 @@ static int stm32_sai_probe(struct platform_device *pdev)
 	}
 
 	/* reset */
-	rst = reset_control_get(&pdev->dev, NULL);
+	rst = reset_control_get_exclusive(&pdev->dev, NULL);
 	if (!IS_ERR(rst)) {
 		reset_control_assert(rst);
 		udelay(2);
diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c
index 4e4250bdb75a..84cc5678beba 100644
--- a/sound/soc/stm/stm32_spdifrx.c
+++ b/sound/soc/stm/stm32_spdifrx.c
@@ -930,7 +930,7 @@ static int stm32_spdifrx_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	rst = devm_reset_control_get(&pdev->dev, NULL);
+	rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
 	if (!IS_ERR(rst)) {
 		reset_control_assert(rst);
 		udelay(2);
-- 
cgit v1.2.3


From 72bfa2117bdb4bb5b07dd5ed833ff3c318fc70b6 Mon Sep 17 00:00:00 2001
From: Philipp Zabel
Date: Wed, 19 Jul 2017 17:26:43 +0200
Subject: ASoC: sun4i: explicitly request exclusive reset control

Commit a53e35db70d1 ("reset: Ensure drivers are explicit when requesting
reset lines") started to transition the reset control request API calls
to explicitly state whether the driver needs exclusive or shared reset
control behavior. Convert all drivers requesting exclusive resets to the
explicit API call so the temporary transition helpers can be removed.

No functional changes.

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Maxime Ripard <maxime.ripard@free-electrons.com>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-codec.c | 3 ++-
 sound/soc/sunxi/sun4i-i2s.c   | 2 +-
 sound/soc/sunxi/sun4i-spdif.c | 3 ++-
 3 files changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 4c37231f254b..73d054f466c2 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -1573,7 +1573,8 @@ static int sun4i_codec_probe(struct platform_device *pdev)
 	}
 
 	if (quirks->has_reset) {
-		scodec->rst = devm_reset_control_get(&pdev->dev, NULL);
+		scodec->rst = devm_reset_control_get_exclusive(&pdev->dev,
+							       NULL);
 		if (IS_ERR(scodec->rst)) {
 			dev_err(&pdev->dev, "Failed to get reset control\n");
 			return PTR_ERR(scodec->rst);
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 3635bbc72cbc..62b307b0c846 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -716,7 +716,7 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 	}
 
 	if (quirks->has_reset) {
-		i2s->rst = devm_reset_control_get(&pdev->dev, NULL);
+		i2s->rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
 		if (IS_ERR(i2s->rst)) {
 			dev_err(&pdev->dev, "Failed to get reset control\n");
 			return PTR_ERR(i2s->rst);
diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c
index eaefd07a5ed0..c49f3757b686 100644
--- a/sound/soc/sunxi/sun4i-spdif.c
+++ b/sound/soc/sunxi/sun4i-spdif.c
@@ -520,7 +520,8 @@ static int sun4i_spdif_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, host);
 
 	if (quirks->has_reset) {
-		host->rst = devm_reset_control_get_optional(&pdev->dev, NULL);
+		host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev,
+								      NULL);
 		if (IS_ERR(host->rst) && PTR_ERR(host->rst) == -EPROBE_DEFER) {
 			ret = -EPROBE_DEFER;
 			dev_err(&pdev->dev, "Failed to get reset: %d\n", ret);
-- 
cgit v1.2.3


From 181e8ce6a6c9a25ffbf10b09d8481f134c5a79ba Mon Sep 17 00:00:00 2001
From: Philipp Zabel
Date: Wed, 19 Jul 2017 17:26:44 +0200
Subject: ASoC: tegra: explicitly request exclusive reset control

Commit a53e35db70d1 ("reset: Ensure drivers are explicit when requesting
reset lines") started to transition the reset control request API calls
to explicitly state whether the driver needs exclusive or shared reset
control behavior. Convert all drivers requesting exclusive resets to the
explicit API call so the temporary transition helpers can be removed.

No functional changes.

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Thierry Reding <thierry.reding@gmail.com>
Cc: Jonathan Hunter <jonathanh@nvidia.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-tegra@vger.kernel.org
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra30_ahub.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
index 8c10ae7982ba..43679aeeb12b 100644
--- a/sound/soc/tegra/tegra30_ahub.c
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -544,8 +544,8 @@ static int tegra30_ahub_probe(struct platform_device *pdev)
 					soc_data->mod_list_mask))
 			continue;
 
-		rst = reset_control_get(&pdev->dev,
-					configlink_mods[i].rst_name);
+		rst = reset_control_get_exclusive(&pdev->dev,
+						  configlink_mods[i].rst_name);
 		if (IS_ERR(rst)) {
 			dev_err(&pdev->dev, "Can't get reset %s\n",
 				configlink_mods[i].rst_name);
-- 
cgit v1.2.3


From 0e46ccd5bc12ab28da9d2f514492aeead65f8332 Mon Sep 17 00:00:00 2001
From: Pradeep Tewani
Date: Thu, 20 Jul 2017 11:40:55 +0530
Subject: ASoC: Intel: board: Remove .owner initialization in bxt_rt298 driver

The .owner field is not required to be initialized in the driver. So,
remove it.

Signed-off-by: Pradeep Tewani <pradeep.d.tewani@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/bxt_rt298.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index 0c3a3cbcb884..1daf98ef9d28 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -492,7 +492,6 @@ static int bxt_card_late_probe(struct snd_soc_card *card)
 /* broxton audio machine driver for SPT + RT298S */
 static struct snd_soc_card broxton_rt298 = {
 	.name = "broxton-rt298",
-	.owner = THIS_MODULE,
 	.dai_link = broxton_rt298_dais,
 	.num_links = ARRAY_SIZE(broxton_rt298_dais),
 	.controls = broxton_controls,
-- 
cgit v1.2.3


From 5622bc955898f4d5fa2ce8f507e5991b08367824 Mon Sep 17 00:00:00 2001
From: Pradeep Tewani
Date: Thu, 20 Jul 2017 11:40:56 +0530
Subject: ASoC: hdac_hdmi: Add the vendor nid for Geminilake HDMI

Geminilake HDMI codec has the different vendor nid. This patch adds the
vendor nid as the part of driver data and uses it.

Signed-off-by: Pradeep Tewani <pradeep.d.tewani@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdac_hdmi.c | 39 ++++++++++++++++++++++++++++++++++-----
 1 file changed, 34 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index bc2e74ff3b2d..eb3ed8c5ed30 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -121,6 +121,10 @@ struct hdac_hdmi_dai_port_map {
 	struct hdac_hdmi_cvt *cvt;
 };
 
+struct hdac_hdmi_drv_data {
+	unsigned int vendor_nid;
+};
+
 struct hdac_hdmi_priv {
 	struct hdac_hdmi_dai_port_map dai_map[HDA_MAX_CVTS];
 	struct list_head pin_list;
@@ -131,6 +135,7 @@ struct hdac_hdmi_priv {
 	int num_ports;
 	struct mutex pin_mutex;
 	struct hdac_chmap chmap;
+	struct hdac_hdmi_drv_data *drv_data;
 };
 
 static struct hdac_hdmi_pcm *
@@ -1321,6 +1326,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid)
 }
 
 #define INTEL_VENDOR_NID 0x08
+#define INTEL_GLK_VENDOR_NID 0x0b
 #define INTEL_GET_VENDOR_VERB 0xf81
 #define INTEL_SET_VENDOR_VERB 0x781
 #define INTEL_EN_DP12			0x02 /* enable DP 1.2 features */
@@ -1329,14 +1335,17 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid)
 static void hdac_hdmi_skl_enable_all_pins(struct hdac_device *hdac)
 {
 	unsigned int vendor_param;
+	struct hdac_ext_device *edev = to_ehdac_device(hdac);
+	struct hdac_hdmi_priv *hdmi = edev->private_data;
+	unsigned int vendor_nid = hdmi->drv_data->vendor_nid;
 
-	vendor_param = snd_hdac_codec_read(hdac, INTEL_VENDOR_NID, 0,
+	vendor_param = snd_hdac_codec_read(hdac, vendor_nid, 0,
 				INTEL_GET_VENDOR_VERB, 0);
 	if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS)
 		return;
 
 	vendor_param |= INTEL_EN_ALL_PIN_CVTS;
-	vendor_param = snd_hdac_codec_read(hdac, INTEL_VENDOR_NID, 0,
+	vendor_param = snd_hdac_codec_read(hdac, vendor_nid, 0,
 				INTEL_SET_VENDOR_VERB, vendor_param);
 	if (vendor_param == -1)
 		return;
@@ -1345,15 +1354,18 @@ static void hdac_hdmi_skl_enable_all_pins(struct hdac_device *hdac)
 static void hdac_hdmi_skl_enable_dp12(struct hdac_device *hdac)
 {
 	unsigned int vendor_param;
+	struct hdac_ext_device *edev = to_ehdac_device(hdac);
+	struct hdac_hdmi_priv *hdmi = edev->private_data;
+	unsigned int vendor_nid = hdmi->drv_data->vendor_nid;
 
-	vendor_param = snd_hdac_codec_read(hdac, INTEL_VENDOR_NID, 0,
+	vendor_param = snd_hdac_codec_read(hdac, vendor_nid, 0,
 				INTEL_GET_VENDOR_VERB, 0);
 	if (vendor_param == -1 || vendor_param & INTEL_EN_DP12)
 		return;
 
 	/* enable DP1.2 mode */
 	vendor_param |= INTEL_EN_DP12;
-	vendor_param = snd_hdac_codec_read(hdac, INTEL_VENDOR_NID, 0,
+	vendor_param = snd_hdac_codec_read(hdac, vendor_nid, 0,
 				INTEL_SET_VENDOR_VERB, vendor_param);
 	if (vendor_param == -1)
 		return;
@@ -1927,6 +1939,14 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx)
 	return port->eld.info.spk_alloc;
 }
 
+static struct hdac_hdmi_drv_data intel_glk_drv_data  = {
+	.vendor_nid = INTEL_GLK_VENDOR_NID,
+};
+
+static struct hdac_hdmi_drv_data intel_drv_data  = {
+	.vendor_nid = INTEL_VENDOR_NID,
+};
+
 static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 {
 	struct hdac_device *codec = &edev->hdac;
@@ -1935,6 +1955,8 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 	struct hdac_ext_link *hlink = NULL;
 	int num_dais = 0;
 	int ret = 0;
+	struct hdac_driver *hdrv = drv_to_hdac_driver(codec->dev.driver);
+	const struct hda_device_id *hdac_id = hdac_get_device_id(codec, hdrv);
 
 	/* hold the ref while we probe */
 	hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
@@ -1956,6 +1978,12 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 	hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached;
 	hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc;
 
+	if (hdac_id->driver_data)
+		hdmi_priv->drv_data =
+			(struct hdac_hdmi_drv_data *)hdac_id->driver_data;
+	else
+		hdmi_priv->drv_data = &intel_drv_data;
+
 	dev_set_drvdata(&codec->dev, edev);
 
 	INIT_LIST_HEAD(&hdmi_priv->pin_list);
@@ -2127,7 +2155,8 @@ static const struct hda_device_id hdmi_list[] = {
 	HDA_CODEC_EXT_ENTRY(0x80862809, 0x100000, "Skylake HDMI", 0),
 	HDA_CODEC_EXT_ENTRY(0x8086280a, 0x100000, "Broxton HDMI", 0),
 	HDA_CODEC_EXT_ENTRY(0x8086280b, 0x100000, "Kabylake HDMI", 0),
-	HDA_CODEC_EXT_ENTRY(0x8086280d, 0x100000, "Geminilake HDMI", 0),
+	HDA_CODEC_EXT_ENTRY(0x8086280d, 0x100000, "Geminilake HDMI",
+						   &intel_glk_drv_data),
 	{}
 };
 
-- 
cgit v1.2.3


From 669eb871c440e19351c2d476d1837ed8a62afe16 Mon Sep 17 00:00:00 2001
From: Pradeep Tewani
Date: Thu, 20 Jul 2017 11:40:57 +0530
Subject: ASoC: Intel: board: Add Geminilake platform support

Geminilake also features rt298 codec, so use the same machine driver
as Broxton. Geminilake uses SSP2 instead of SSP5.

Signed-off-by: Pradeep Tewani <pradeep.d.tewani@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/bxt_rt298.c | 84 ++++++++++++++++++++++++++++++++++++--
 1 file changed, 81 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index 1daf98ef9d28..a4b5bcb1f6f8 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -114,7 +114,44 @@ static const struct snd_soc_dapm_route broxton_rt298_map[] = {
 	{ "iDisp2 Tx", NULL, "iDisp2_out"},
 	{ "hifi1", NULL, "iDisp1 Tx"},
 	{ "iDisp1 Tx", NULL, "iDisp1_out"},
+};
+
+static const struct snd_soc_dapm_route geminilake_rt298_map[] = {
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack detect */
+	{"Headphone Jack", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC2"},
+	{"DMic", NULL, "SoC DMIC"},
+
+	{"HDMI1", NULL, "hif5-0 Output"},
+	{"HDMI2", NULL, "hif6-0 Output"},
+	{"HDMI2", NULL, "hif7-0 Output"},
+
+	/* CODEC BE connections */
+	{ "AIF1 Playback", NULL, "ssp2 Tx"},
+	{ "ssp2 Tx", NULL, "codec0_out"},
+	{ "ssp2 Tx", NULL, "codec1_out"},
+
+	{ "codec0_in", NULL, "ssp2 Rx" },
+	{ "ssp2 Rx", NULL, "AIF1 Capture" },
 
+	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
+	{ "DMIC01 Rx", NULL, "Capture" },
+
+	{ "hifi3", NULL, "iDisp3 Tx"},
+	{ "iDisp3 Tx", NULL, "iDisp3_out"},
+	{ "hifi2", NULL, "iDisp2 Tx"},
+	{ "iDisp2 Tx", NULL, "iDisp2_out"},
+	{ "hifi1", NULL, "iDisp1 Tx"},
+	{ "iDisp1 Tx", NULL, "iDisp1_out"},
 };
 
 static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd)
@@ -505,9 +542,41 @@ static struct snd_soc_card broxton_rt298 = {
 
 };
 
+static struct snd_soc_card geminilake_rt298 = {
+	.name = "geminilake-rt298",
+	.dai_link = broxton_rt298_dais,
+	.num_links = ARRAY_SIZE(broxton_rt298_dais),
+	.controls = broxton_controls,
+	.num_controls = ARRAY_SIZE(broxton_controls),
+	.dapm_widgets = broxton_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(broxton_widgets),
+	.dapm_routes = geminilake_rt298_map,
+	.num_dapm_routes = ARRAY_SIZE(geminilake_rt298_map),
+	.fully_routed = true,
+	.late_probe = bxt_card_late_probe,
+};
+
 static int broxton_audio_probe(struct platform_device *pdev)
 {
 	struct bxt_rt286_private *ctx;
+	struct snd_soc_card *card =
+			(struct snd_soc_card *)pdev->id_entry->driver_data;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) {
+		if (!strncmp(card->dai_link[i].codec_name, "i2c-INT343A:00",
+						I2C_NAME_SIZE)) {
+			if (!strncmp(card->name, "broxton-rt298",
+						PLATFORM_NAME_SIZE)) {
+				card->dai_link[i].name = "SSP5-Codec";
+				card->dai_link[i].cpu_dai_name = "SSP5 Pin";
+			} else if (!strncmp(card->name, "geminilake-rt298",
+						PLATFORM_NAME_SIZE)) {
+				card->dai_link[i].name = "SSP2-Codec";
+				card->dai_link[i].cpu_dai_name = "SSP2 Pin";
+			}
+		}
+	}
 
 	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
 	if (!ctx)
@@ -515,18 +584,26 @@ static int broxton_audio_probe(struct platform_device *pdev)
 
 	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
 
-	broxton_rt298.dev = &pdev->dev;
-	snd_soc_card_set_drvdata(&broxton_rt298, ctx);
+	card->dev = &pdev->dev;
+	snd_soc_card_set_drvdata(card, ctx);
 
-	return devm_snd_soc_register_card(&pdev->dev, &broxton_rt298);
+	return devm_snd_soc_register_card(&pdev->dev, card);
 }
 
+static const struct platform_device_id bxt_board_ids[] = {
+	{ .name = "bxt_alc298s_i2s", .driver_data =
+				(unsigned long)&broxton_rt298 },
+	{ .name = "glk_alc298s_i2s", .driver_data =
+				(unsigned long)&geminilake_rt298 },
+};
+
 static struct platform_driver broxton_audio = {
 	.probe = broxton_audio_probe,
 	.driver = {
 		.name = "bxt_alc298s_i2s",
 		.pm = &snd_soc_pm_ops,
 	},
+	.id_table = bxt_board_ids,
 };
 module_platform_driver(broxton_audio)
 
@@ -536,3 +613,4 @@ MODULE_AUTHOR("Senthilnathan Veppur <senthilnathanx.veppur@intel.com>");
 MODULE_DESCRIPTION("Intel SST Audio for Broxton");
 MODULE_LICENSE("GPL v2");
 MODULE_ALIAS("platform:bxt_alc298s_i2s");
+MODULE_ALIAS("platform:glk_alc298s_i2s");
-- 
cgit v1.2.3


From c0d1cb8366bab9963822c27b0d40cb8b32928cdc Mon Sep 17 00:00:00 2001
From: Andy Shevchenko
Date: Tue, 18 Jul 2017 20:34:19 +0300
Subject: ASoC: rt5677: Remove never used variables

There are two variables that have never been used.

sound/soc/codecs/rt5677.c:785:35: warning: ‘in_vol_tlv’ defined but not used [-Wunused-const-variable=]
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);

sound/soc/codecs/rt5677.c:783:35: warning: ‘out_vol_tlv’ defined but not used [-Wunused-const-variable=]
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);

Simple remove them.

Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index a3cce57739c3..9e9de0e274ea 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -780,9 +780,7 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on)
 	return 0;
 }
 
-static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
 static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0);
-static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
 static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
 static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0);
-- 
cgit v1.2.3


From e8be3a5ad335316ddf1b90e0386426093187bfc7 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Thu, 20 Jul 2017 12:05:34 +0800
Subject: ASoC: rt5514: Support the TDM docking mode

The TDM docking mode can share the bus to other devices simultaneously.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 33 ++++++++++++++++++++++++++++++++-
 sound/soc/codecs/rt5514.h | 12 ++++++++++++
 2 files changed, 44 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 4ca73bf83cc1..a105c84ef555 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -871,11 +871,38 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec);
-	unsigned int val = 0;
+	unsigned int val = 0, val2 = 0;
 
 	if (rx_mask || tx_mask)
 		val |= RT5514_TDM_MODE;
 
+	switch (tx_mask) {
+	case 0x3:
+		val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 |
+			RT5514_TDM_DOCKING_START_SLOT0;
+		break;
+
+	case 0x30:
+		val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 |
+			RT5514_TDM_DOCKING_START_SLOT4;
+		break;
+
+	case 0xf:
+		val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 |
+			RT5514_TDM_DOCKING_START_SLOT0;
+		break;
+
+	case 0xf0:
+		val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 |
+			RT5514_TDM_DOCKING_START_SLOT4;
+		break;
+
+	default:
+		break;
+	}
+
+
+
 	switch (slots) {
 	case 4:
 		val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH;
@@ -921,6 +948,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK |
 		RT5514_TDM_MODE2, val);
 
+	regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL2,
+		RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH_MASK |
+		RT5514_TDM_DOCKING_START_MASK, val2);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h
index 02bc212a86d9..75024a4bb7f5 100644
--- a/sound/soc/codecs/rt5514.h
+++ b/sound/soc/codecs/rt5514.h
@@ -164,6 +164,18 @@
 #define RT5514_I2S_DL_24			(0x2 << 0)
 #define RT5514_I2S_DL_8				(0x3 << 0)
 
+/* RT5514_I2S_CTRL2 (0x2014) */
+#define RT5514_TDM_DOCKING_MODE			(0x1 << 31)
+#define RT5514_TDM_DOCKING_MODE_SFT		31
+#define RT5514_TDM_DOCKING_VALID_CH_MASK	(0x1 << 29)
+#define RT5514_TDM_DOCKING_VALID_CH_SFT		29
+#define RT5514_TDM_DOCKING_VALID_CH2		(0x0 << 29)
+#define RT5514_TDM_DOCKING_VALID_CH4		(0x1 << 29)
+#define RT5514_TDM_DOCKING_START_MASK		(0x1 << 28)
+#define RT5514_TDM_DOCKING_START_SFT		28
+#define RT5514_TDM_DOCKING_START_SLOT0		(0x0 << 28)
+#define RT5514_TDM_DOCKING_START_SLOT4		(0x1 << 28)
+
 /* RT5514_DIG_SOURCE_CTRL (0x20a4) */
 #define RT5514_AD1_DMIC_INPUT_SEL		(0x1 << 1)
 #define RT5514_AD1_DMIC_INPUT_SEL_SFT		1
-- 
cgit v1.2.3


From 01dfb1ec15ce6120055401953265c7b51e899292 Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Thu, 20 Jul 2017 13:07:50 +0800
Subject: ASoC: rt5665: add clcok control for master mode

Add i2s clock control for codec master mode.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5665.c | 24 +++++++++++++++++++++++-
 sound/soc/codecs/rt5665.h | 21 +++++++++++++++++++++
 2 files changed, 44 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index d3103efcb135..ef27561c4993 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -4186,6 +4186,15 @@ static int rt5665_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
+	if (rt5665->master[RT5665_AIF2_1] || rt5665->master[RT5665_AIF2_2]) {
+		snd_soc_update_bits(codec, RT5665_I2S_M_CLK_CTRL_1,
+			RT5665_I2S2_M_PD_MASK, pre_div << RT5665_I2S2_M_PD_SFT);
+	}
+	if (rt5665->master[RT5665_AIF3]) {
+		snd_soc_update_bits(codec, RT5665_I2S_M_CLK_CTRL_1,
+			RT5665_I2S3_M_PD_MASK, pre_div << RT5665_I2S3_M_PD_SFT);
+	}
+
 	return 0;
 }
 
@@ -4262,7 +4271,7 @@ static int rt5665_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id,
 				   int source, unsigned int freq, int dir)
 {
 	struct rt5665_priv *rt5665 = snd_soc_codec_get_drvdata(codec);
-	unsigned int reg_val = 0;
+	unsigned int reg_val = 0, src = 0;
 
 	if (freq == rt5665->sysclk && clk_id == rt5665->sysclk_src)
 		return 0;
@@ -4270,12 +4279,15 @@ static int rt5665_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id,
 	switch (clk_id) {
 	case RT5665_SCLK_S_MCLK:
 		reg_val |= RT5665_SCLK_SRC_MCLK;
+		src = RT5665_CLK_SRC_MCLK;
 		break;
 	case RT5665_SCLK_S_PLL1:
 		reg_val |= RT5665_SCLK_SRC_PLL1;
+		src = RT5665_CLK_SRC_PLL1;
 		break;
 	case RT5665_SCLK_S_RCCLK:
 		reg_val |= RT5665_SCLK_SRC_RCCLK;
+		src = RT5665_CLK_SRC_RCCLK;
 		break;
 	default:
 		dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id);
@@ -4283,6 +4295,16 @@ static int rt5665_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id,
 	}
 	snd_soc_update_bits(codec, RT5665_GLB_CLK,
 		RT5665_SCLK_SRC_MASK, reg_val);
+
+	if (rt5665->master[RT5665_AIF2_1] || rt5665->master[RT5665_AIF2_2]) {
+		snd_soc_update_bits(codec, RT5665_I2S_M_CLK_CTRL_1,
+			RT5665_I2S2_SRC_MASK, src << RT5665_I2S2_SRC_SFT);
+	}
+	if (rt5665->master[RT5665_AIF3]) {
+		snd_soc_update_bits(codec, RT5665_I2S_M_CLK_CTRL_1,
+			RT5665_I2S3_SRC_MASK, src << RT5665_I2S3_SRC_SFT);
+	}
+
 	rt5665->sysclk = freq;
 	rt5665->sysclk_src = clk_id;
 
diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h
index 1db5c6a62a8e..8f08acb9c446 100644
--- a/sound/soc/codecs/rt5665.h
+++ b/sound/soc/codecs/rt5665.h
@@ -1628,6 +1628,27 @@
 #define RT5665_PWR_CLK1M_PD			(0x0 << 8)
 #define RT5665_PWR_CLK1M_PU			(0x1 << 8)
 
+/* I2S Master Mode Clock Control 1 (0x00a0) */
+#define RT5665_CLK_SRC_MCLK			(0x0)
+#define RT5665_CLK_SRC_PLL1			(0x1)
+#define RT5665_CLK_SRC_RCCLK			(0x2)
+#define RT5665_I2S_PD_1				(0x0)
+#define RT5665_I2S_PD_2				(0x1)
+#define RT5665_I2S_PD_3				(0x2)
+#define RT5665_I2S_PD_4				(0x3)
+#define RT5665_I2S_PD_6				(0x4)
+#define RT5665_I2S_PD_8				(0x5)
+#define RT5665_I2S_PD_12			(0x6)
+#define RT5665_I2S_PD_16			(0x7)
+#define RT5665_I2S2_SRC_MASK			(0x3 << 12)
+#define RT5665_I2S2_SRC_SFT			12
+#define RT5665_I2S2_M_PD_MASK			(0x7 << 8)
+#define RT5665_I2S2_M_PD_SFT			8
+#define RT5665_I2S3_SRC_MASK			(0x3 << 4)
+#define RT5665_I2S3_SRC_SFT			4
+#define RT5665_I2S3_M_PD_MASK			(0x7 << 0)
+#define RT5665_I2S3_M_PD_SFT			0
+
 
 /* EQ Control 1 (0x00b0) */
 #define RT5665_EQ_SRC_DAC			(0x0 << 15)
-- 
cgit v1.2.3


From 8bfa426845419d1fe895fc5668ab29b3a0c36c85 Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Thu, 20 Jul 2017 13:07:51 +0800
Subject: ASoC: rt5665: add clock sync control for master mode

We can sleect the i2s clock source for each i2s bus in master mode.
The choice is depended on the HW design and the use case.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5665.c | 15 +++++++++++++++
 1 file changed, 15 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index ef27561c4993..7e64b25f8585 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -1381,6 +1381,16 @@ static void rt5665_jack_detect_handler(struct work_struct *work)
 	mutex_unlock(&rt5665->calibrate_mutex);
 }
 
+static const char * const rt5665_clk_sync[] = {
+	"I2S1_1", "I2S1_2", "I2S2", "I2S3", "IF2 Slave", "IF3 Slave"
+};
+
+static const struct soc_enum rt5665_enum[] = {
+	SOC_ENUM_SINGLE(RT5665_I2S1_SDP, 11, 5, rt5665_clk_sync),
+	SOC_ENUM_SINGLE(RT5665_I2S2_SDP, 11, 5, rt5665_clk_sync),
+	SOC_ENUM_SINGLE(RT5665_I2S3_SDP, 11, 5, rt5665_clk_sync),
+};
+
 static const struct snd_kcontrol_new rt5665_snd_controls[] = {
 	/* Headphone Output Volume */
 	SOC_DOUBLE_R_EXT_TLV("Headphone Playback Volume", RT5665_HPL_GAIN,
@@ -1446,6 +1456,11 @@ static const struct snd_kcontrol_new rt5665_snd_controls[] = {
 	SOC_DOUBLE_TLV("STO2 ADC Boost Gain Volume", RT5665_STO2_ADC_BOOST,
 		RT5665_STO2_ADC_L_BST_SFT, RT5665_STO2_ADC_R_BST_SFT,
 		3, 0, adc_bst_tlv),
+
+	/* I2S3 CLK Source */
+	SOC_ENUM("I2S1 Master Clk Sel", rt5665_enum[0]),
+	SOC_ENUM("I2S2 Master Clk Sel", rt5665_enum[1]),
+	SOC_ENUM("I2S3 Master Clk Sel", rt5665_enum[2]),
 };
 
 /**
-- 
cgit v1.2.3


From b833786b69c14fe98846c38d36d8f6e5e92ccdd8 Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Mon, 24 Jul 2017 19:11:52 +0800
Subject: ASoC: rt5665: add MONOVOL Playback Volume control

MONOVOL Playback Volume is a playback volume control on "MONOVOL MIX".

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5665.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index 7e64b25f8585..54ace3b98ec3 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -1402,6 +1402,9 @@ static const struct snd_kcontrol_new rt5665_snd_controls[] = {
 		RT5665_L_VOL_SFT, 15, 1, snd_soc_get_volsw,
 		rt5665_mono_vol_put, mono_vol_tlv),
 
+	SOC_SINGLE_TLV("MONOVOL Playback Volume", RT5665_MONO_OUT,
+		RT5665_L_VOL_SFT, 39, 1, out_vol_tlv),
+
 	/* Output Volume */
 	SOC_DOUBLE_TLV("OUT Playback Volume", RT5665_LOUT, RT5665_L_VOL_SFT,
 		RT5665_R_VOL_SFT, 39, 1, out_vol_tlv),
-- 
cgit v1.2.3


From 16d4498962c7b76a76553e8dd943abe1e9facc0d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Mon, 24 Jul 2017 01:45:59 +0000
Subject: ASoC: rsnd: don't use private_value on rsnd_kctrl_new()

Current rsnd kctrl is using both .private_value (for rsnd_kctrl_cfg)
and .private_data (for rsnd_mod) on snd_kcontrol.
But only 1 private data (= rsnd_kctrl_cfg) can be enough if it has
rsnd_mod pointer.
This patch doesn't use private_value.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/core.c | 14 ++++++--------
 sound/soc/sh/rcar/rsnd.h |  1 +
 2 files changed, 7 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 3f2ced26ed37..acafb838b7b7 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1169,11 +1169,10 @@ static struct snd_pcm_ops rsnd_pcm_ops = {
 /*
  *		snd_kcontrol
  */
-#define kcontrol_to_cfg(kctrl) ((struct rsnd_kctrl_cfg *)kctrl->private_value)
 static int rsnd_kctrl_info(struct snd_kcontrol *kctrl,
 			   struct snd_ctl_elem_info *uinfo)
 {
-	struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl);
+	struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl);
 
 	if (cfg->texts) {
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
@@ -1199,7 +1198,7 @@ static int rsnd_kctrl_info(struct snd_kcontrol *kctrl,
 static int rsnd_kctrl_get(struct snd_kcontrol *kctrl,
 			  struct snd_ctl_elem_value *uc)
 {
-	struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl);
+	struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl);
 	int i;
 
 	for (i = 0; i < cfg->size; i++)
@@ -1214,8 +1213,7 @@ static int rsnd_kctrl_get(struct snd_kcontrol *kctrl,
 static int rsnd_kctrl_put(struct snd_kcontrol *kctrl,
 			  struct snd_ctl_elem_value *uc)
 {
-	struct rsnd_mod *mod = snd_kcontrol_chip(kctrl);
-	struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl);
+	struct rsnd_kctrl_cfg *cfg = snd_kcontrol_chip(kctrl);
 	int i, change = 0;
 
 	if (!cfg->accept(cfg->io))
@@ -1232,7 +1230,7 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl,
 	}
 
 	if (change && cfg->update)
-		cfg->update(cfg->io, mod);
+		cfg->update(cfg->io, cfg->mod);
 
 	return change;
 }
@@ -1284,14 +1282,13 @@ int rsnd_kctrl_new(struct rsnd_mod *mod,
 		.index		= rtd->num,
 		.get		= rsnd_kctrl_get,
 		.put		= rsnd_kctrl_put,
-		.private_value	= (unsigned long)cfg,
 	};
 	int ret;
 
 	if (size > RSND_MAX_CHANNELS)
 		return -EINVAL;
 
-	kctrl = snd_ctl_new1(&knew, mod);
+	kctrl = snd_ctl_new1(&knew, cfg);
 	if (!kctrl)
 		return -ENOMEM;
 
@@ -1307,6 +1304,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod,
 	cfg->card	= card;
 	cfg->kctrl	= kctrl;
 	cfg->io		= io;
+	cfg->mod	= mod;
 
 	return 0;
 }
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 99c57611df88..c5de71f2dc8c 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -614,6 +614,7 @@ struct rsnd_kctrl_cfg {
 	struct rsnd_dai_stream *io;
 	struct snd_card *card;
 	struct snd_kcontrol *kctrl;
+	struct rsnd_mod *mod;
 };
 
 #define RSND_MAX_CHANNELS	8
-- 
cgit v1.2.3


From 1628ef89042fdb90aaea657942455e00e160174d Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Mon, 24 Jul 2017 15:34:23 +0800
Subject: ASoC: rt5514: Add the I2S ASRC support

The patch adds the I2S ASRC support.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 19 +++++++++++++++++++
 sound/soc/codecs/rt5514.h |  7 +++++++
 2 files changed, 26 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index a105c84ef555..882fdc4a6fd4 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -63,6 +63,7 @@ static const struct reg_sequence rt5514_patch[] = {
 	{RT5514_SRC_CTRL,		0x44000eee},
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_ADCFED,	0x00000800},
+	{RT5514_ASRC_IN_CTRL1,		0x00000003},
 };
 
 static const struct reg_default rt5514_reg[] = {
@@ -471,6 +472,15 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
 		return 0;
 }
 
+static int rt5514_i2s_use_asrc(struct snd_soc_dapm_widget *source,
+	struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+	struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec);
+
+	return (rt5514->sysclk > rt5514->lrck * 384);
+}
+
 static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = {
 	/* Input Lines */
 	SND_SOC_DAPM_INPUT("DMIC1L"),
@@ -541,6 +551,10 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = {
 		RT5514_POW_PLL1_LDO_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5514_PWR_ANA2, RT5514_POW_PLL1_BIT, 0,
 		NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("ASRC AD1", 1, RT5514_CLK_CTRL2,
+		RT5514_CLK_AD0_ASRC_EN_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY_S("ASRC AD2", 1, RT5514_CLK_CTRL2,
+		RT5514_CLK_AD1_ASRC_EN_BIT, 0, NULL, 0),
 
 	/* ADC Mux */
 	SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0,
@@ -638,6 +652,7 @@ static const struct snd_soc_dapm_route rt5514_dapm_routes[] = {
 	{ "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" },
 	{ "Stereo1 ADC MIX", NULL, "adc stereo1 filter" },
 	{ "adc stereo1 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll },
+	{ "adc stereo1 filter", NULL, "ASRC AD1", rt5514_i2s_use_asrc },
 
 	{ "Stereo2 DMIC Mux", "DMIC1", "DMIC1" },
 	{ "Stereo2 DMIC Mux", "DMIC2", "DMIC2" },
@@ -654,6 +669,7 @@ static const struct snd_soc_dapm_route rt5514_dapm_routes[] = {
 	{ "Stereo2 ADC MIX", NULL, "Stereo2 ADC MIXR" },
 	{ "Stereo2 ADC MIX", NULL, "adc stereo2 filter" },
 	{ "adc stereo2 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll },
+	{ "adc stereo2 filter", NULL, "ASRC AD2", rt5514_i2s_use_asrc },
 
 	{ "AIF1TX", NULL, "Stereo1 ADC MIX"},
 	{ "AIF1TX", NULL, "Stereo2 ADC MIX"},
@@ -706,6 +722,9 @@ static int rt5514_hw_params(struct snd_pcm_substream *substream,
 
 	regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_I2S_DL_MASK,
 		val_len);
+	regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL1,
+		RT5514_CLK_AD_ANA1_SEL_MASK,
+		(pre_div + 1) << RT5514_CLK_AD_ANA1_SEL_SFT);
 	regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL2,
 		RT5514_CLK_SYS_DIV_OUT_MASK | RT5514_SEL_ADC_OSR_MASK,
 		pre_div << RT5514_CLK_SYS_DIV_OUT_SFT |
diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h
index 75024a4bb7f5..2dc40e6d8b3f 100644
--- a/sound/soc/codecs/rt5514.h
+++ b/sound/soc/codecs/rt5514.h
@@ -37,6 +37,7 @@
 #define RT5514_PLL3_CALIB_CTRL5			0x2124
 #define RT5514_DELAY_BUF_CTRL1			0x2140
 #define RT5514_DELAY_BUF_CTRL3			0x2148
+#define RT5514_ASRC_IN_CTRL1			0x2180
 #define RT5514_DOWNFILTER0_CTRL1		0x2190
 #define RT5514_DOWNFILTER0_CTRL2		0x2194
 #define RT5514_DOWNFILTER0_CTRL3		0x2198
@@ -197,8 +198,14 @@
 #define RT5514_CLK_AD0_EN_BIT			23
 #define RT5514_CLK_DMIC_OUT_SEL_MASK		(0x7 << 8)
 #define RT5514_CLK_DMIC_OUT_SEL_SFT		8
+#define RT5514_CLK_AD_ANA1_SEL_MASK		(0xf << 0)
+#define RT5514_CLK_AD_ANA1_SEL_SFT		0
 
 /* RT5514_CLK_CTRL2 (0x2108) */
+#define RT5514_CLK_AD1_ASRC_EN			(0x1 << 17)
+#define RT5514_CLK_AD1_ASRC_EN_BIT		17
+#define RT5514_CLK_AD0_ASRC_EN			(0x1 << 16)
+#define RT5514_CLK_AD0_ASRC_EN_BIT		16
 #define RT5514_CLK_SYS_DIV_OUT_MASK		(0x7 << 8)
 #define RT5514_CLK_SYS_DIV_OUT_SFT		8
 #define RT5514_SEL_ADC_OSR_MASK			(0x7 << 4)
-- 
cgit v1.2.3


From 8a43c2240c862bbf3fc612d0533b891bd75ec771 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Sun, 23 Jul 2017 22:53:24 +0530
Subject: ASoC: rt5514: constify acpi_device_id.

acpi_device_id are not supposed to change at runtime. All functions
working with acpi_device_id provided by <acpi/acpi_bus.h> work with
const acpi_device_id. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 882fdc4a6fd4..3d0aea8f3801 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -1134,7 +1134,7 @@ MODULE_DEVICE_TABLE(of, rt5514_of_match);
 #endif
 
 #ifdef CONFIG_ACPI
-static struct acpi_device_id rt5514_acpi_match[] = {
+static const struct acpi_device_id rt5514_acpi_match[] = {
 	{ "10EC5514", 0},
 	{},
 };
-- 
cgit v1.2.3


From 1d89147cc3dfef185e701baad437fa91c0977384 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Sun, 23 Jul 2017 22:53:25 +0530
Subject: ASoC: rt5663: constify acpi_device_id.

acpi_device_id are not supposed to change at runtime. All functions
working with acpi_device_id provided by <acpi/acpi_bus.h> work with
const acpi_device_id. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index a33202affeb1..271cc81f2aef 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -2956,7 +2956,7 @@ MODULE_DEVICE_TABLE(of, rt5663_of_match);
 #endif
 
 #ifdef CONFIG_ACPI
-static struct acpi_device_id rt5663_acpi_match[] = {
+static const struct acpi_device_id rt5663_acpi_match[] = {
 	{ "10EC5663", 0},
 	{},
 };
-- 
cgit v1.2.3


From f36544a425d3b75281a1601fe663086aec5b36c7 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Sun, 23 Jul 2017 22:53:26 +0530
Subject: ASoC: rt5659: constify acpi_device_id.

acpi_device_id are not supposed to change at runtime. All functions
working with acpi_device_id provided by <acpi/acpi_bus.h> work with
const acpi_device_id. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5659.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 1b7060850340..0a99fad48392 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -4222,7 +4222,7 @@ MODULE_DEVICE_TABLE(of, rt5659_of_match);
 #endif
 
 #ifdef CONFIG_ACPI
-static struct acpi_device_id rt5659_acpi_match[] = {
+static const struct acpi_device_id rt5659_acpi_match[] = {
 	{ "10EC5658", 0, },
 	{ "10EC5659", 0, },
 	{ },
-- 
cgit v1.2.3


From abfedab1163a6b01ab400a67f4eed0fea3ccbdd1 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Sun, 23 Jul 2017 22:53:27 +0530
Subject: ASoC: rt5665: constify acpi_device_id.

acpi_device_id are not supposed to change at runtime. All functions
working with acpi_device_id provided by <acpi/acpi_bus.h> work with
const acpi_device_id. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5665.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index 54ace3b98ec3..d0a9490f97f1 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -4970,7 +4970,7 @@ MODULE_DEVICE_TABLE(of, rt5665_of_match);
 #endif
 
 #ifdef CONFIG_ACPI
-static struct acpi_device_id rt5665_acpi_match[] = {
+static const struct acpi_device_id rt5665_acpi_match[] = {
 	{"10EC5665", 0,},
 	{"10EC5666", 0,},
 	{"10EC5668", 0,},
-- 
cgit v1.2.3


From c2bf252536577db77c719f62a54cad2e4be5c4d7 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Wed, 26 Jul 2017 18:32:20 +0800
Subject: ASoC: rt5514: Use the IS_ENABLED to supports the module build

The patch uses the macro IS_ENABLED to make sure that the SPI function also
supports the module build.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 3d0aea8f3801..21cbb566c0d3 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -31,7 +31,7 @@
 
 #include "rl6231.h"
 #include "rt5514.h"
-#if defined(CONFIG_SND_SOC_RT5514_SPI)
+#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI)
 #include "rt5514-spi.h"
 #endif
 
@@ -312,7 +312,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 
 			request_firmware(&fw, RT5514_FIRMWARE1, codec->dev);
 			if (fw) {
-#if defined(CONFIG_SND_SOC_RT5514_SPI)
+#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI)
 				rt5514_spi_burst_write(0x4ff60000, fw->data,
 					((fw->size/8)+1)*8);
 #else
@@ -325,7 +325,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 
 			request_firmware(&fw, RT5514_FIRMWARE2, codec->dev);
 			if (fw) {
-#if defined(CONFIG_SND_SOC_RT5514_SPI)
+#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI)
 				rt5514_spi_burst_write(0x4ffc0000, fw->data,
 					((fw->size/8)+1)*8);
 #else
-- 
cgit v1.2.3


From 8215313c8bf487b6b15839be3c18fdd1598b1e79 Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Wed, 26 Jul 2017 15:28:26 +0800
Subject: ASoC: rt5670: add symmetric_rates flag

rt5670 requires symmetric rate of capture and playback.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 0ec7985ed306..174141fefc79 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2743,6 +2743,7 @@ static struct snd_soc_dai_driver rt5670_dai[] = {
 			.formats = RT5670_FORMATS,
 		},
 		.ops = &rt5670_aif_dai_ops,
+		.symmetric_rates = 1,
 	},
 	{
 		.name = "rt5670-aif2",
@@ -2762,6 +2763,7 @@ static struct snd_soc_dai_driver rt5670_dai[] = {
 			.formats = RT5670_FORMATS,
 		},
 		.ops = &rt5670_aif_dai_ops,
+		.symmetric_rates = 1,
 	},
 };
 
-- 
cgit v1.2.3


From a7e1149b64091ac87dfb18b177abb63ba4cbbd3d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Wed, 26 Jul 2017 05:17:08 +0000
Subject: ASoC: remove cache_bypass from snd_soc_codec

snd_soc_codec .cache_bypass related operation code has been removed.
Let's remove remaining code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h         | 1 -
 sound/soc/codecs/max98927.c | 1 -
 2 files changed, 2 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9c94b97c17f8..10b71beb9f13 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -894,7 +894,6 @@ struct snd_soc_codec {
 	struct list_head list;
 
 	/* runtime */
-	unsigned int cache_bypass:1; /* Suppress access to the cache */
 	unsigned int cache_init:1; /* codec cache has been initialized */
 
 	/* codec IO */
diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index b5ee29499e16..b0380b560bf5 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -577,7 +577,6 @@ static int max98927_probe(struct snd_soc_codec *codec)
 
 	max98927->codec = codec;
 	codec->control_data = max98927->regmap;
-	codec->cache_bypass = 1;
 
 	/* Software Reset */
 	regmap_write(max98927->regmap,
-- 
cgit v1.2.3


From d2d2c0a0f0e230e9650e778d0d5c7ed6d1bcffb0 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:44:28 +0530
Subject: ASoC: samsung: s3c2412: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/s3c2412-i2s.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 0a4718207e6e..bcd1cbdeac93 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -71,7 +71,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
 	/* Set MPLL as the source for IIS CLK */
 
 	clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
-	clk_prepare_enable(s3c2412_i2s.iis_cclk);
+	ret = clk_prepare_enable(s3c2412_i2s.iis_cclk);
+	if (ret)
+		return ret;
 
 	s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
 
-- 
cgit v1.2.3


From 8698475a94d0b7eb21efaa346d507dc4932f2032 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:44:29 +0530
Subject: ASoC: samsung: s3c24xx: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/s3c24xx-i2s.c | 11 +++++++++--
 1 file changed, 9 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 91e6871e5413..8d58d02183bf 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -340,6 +340,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
 
 static int s3c24xx_i2s_probe(struct snd_soc_dai *dai)
 {
+	int ret;
 	snd_soc_dai_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out,
 					&s3c24xx_i2s_pcm_stereo_in);
 
@@ -348,7 +349,9 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai)
 		pr_err("failed to get iis_clock\n");
 		return PTR_ERR(s3c24xx_i2s.iis_clk);
 	}
-	clk_prepare_enable(s3c24xx_i2s.iis_clk);
+	ret = clk_prepare_enable(s3c24xx_i2s.iis_clk);
+	if (ret)
+		return ret;
 
 	/* Configure the I2S pins (GPE0...GPE4) in correct mode */
 	s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
@@ -377,7 +380,11 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 
 static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
-	clk_prepare_enable(s3c24xx_i2s.iis_clk);
+	int ret;
+
+	ret = clk_prepare_enable(s3c24xx_i2s.iis_clk);
+	if (ret)
+		return ret;
 
 	writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
 	writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-- 
cgit v1.2.3


From a7ad9a88300c813a945793e9dae84ceaf7809141 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:44:30 +0530
Subject: ASoC: samsung: pcm: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/pcm.c | 8 ++++++--
 1 file changed, 6 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index d50a6377c23d..37f95eee1558 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -522,7 +522,9 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "failed to get audio-bus clock\n");
 		return PTR_ERR(pcm->cclk);
 	}
-	clk_prepare_enable(pcm->cclk);
+	ret = clk_prepare_enable(pcm->cclk);
+	if (ret)
+		return ret;
 
 	/* record our pcm structure for later use in the callbacks */
 	dev_set_drvdata(&pdev->dev, pcm);
@@ -533,7 +535,9 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev)
 		ret = PTR_ERR(pcm->pclk);
 		goto err_dis_cclk;
 	}
-	clk_prepare_enable(pcm->pclk);
+	ret = clk_prepare_enable(pcm->pclk);
+	if (ret)
+		goto err_dis_cclk;
 
 	s3c_pcm_stereo_in[pdev->id].addr = mem_res->start + S3C_PCM_RXFIFO;
 	s3c_pcm_stereo_out[pdev->id].addr = mem_res->start + S3C_PCM_TXFIFO;
-- 
cgit v1.2.3


From f5c97c7b0438bec0aad9738af1baa698dae5bc09 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:44:31 +0530
Subject: ASoC: samsung: i2s: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/i2s.c | 19 +++++++++++++++----
 1 file changed, 15 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index c9f87f7bae99..daf7b892c967 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -551,7 +551,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
 				goto err;
 			}
 
-			clk_prepare_enable(i2s->op_clk);
+			ret = clk_prepare_enable(i2s->op_clk);
+			if (ret)
+				goto err;
 			i2s->rclk_srcrate = clk_get_rate(i2s->op_clk);
 
 			/* Over-ride the other's */
@@ -1133,10 +1135,19 @@ static int i2s_runtime_suspend(struct device *dev)
 static int i2s_runtime_resume(struct device *dev)
 {
 	struct i2s_dai *i2s = dev_get_drvdata(dev);
+	int ret;
 
-	clk_prepare_enable(i2s->clk);
-	if (i2s->op_clk)
-		clk_prepare_enable(i2s->op_clk);
+	ret = clk_prepare_enable(i2s->clk);
+	if (ret)
+		return ret;
+
+	if (i2s->op_clk) {
+		ret = clk_prepare_enable(i2s->op_clk);
+		if (ret) {
+			clk_disable_unprepare(i2s->clk);
+			return ret;
+		}
+	}
 
 	writel(i2s->suspend_i2scon, i2s->addr + I2SCON);
 	writel(i2s->suspend_i2smod, i2s->addr + I2SMOD);
-- 
cgit v1.2.3


From 0f11daf10b4ac6a6c8ea6b39552f119c595b5406 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:44:32 +0530
Subject: ASoC: samsung: spdif: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/spdif.c | 8 ++++++--
 1 file changed, 6 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index 779504f54bc0..cb59911e65c0 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -391,7 +391,9 @@ static int spdif_probe(struct platform_device *pdev)
 		ret = -ENOENT;
 		goto err0;
 	}
-	clk_prepare_enable(spdif->pclk);
+	ret = clk_prepare_enable(spdif->pclk);
+	if (ret)
+		goto err0;
 
 	spdif->sclk = devm_clk_get(&pdev->dev, "sclk_spdif");
 	if (IS_ERR(spdif->sclk)) {
@@ -399,7 +401,9 @@ static int spdif_probe(struct platform_device *pdev)
 		ret = -ENOENT;
 		goto err1;
 	}
-	clk_prepare_enable(spdif->sclk);
+	ret = clk_prepare_enable(spdif->sclk);
+	if (ret)
+		goto err1;
 
 	/* Request S/PDIF Register's memory region */
 	if (!request_mem_region(mem_res->start,
-- 
cgit v1.2.3


From 0d97ee867958a092f02e9cd7c865c278bcec18a6 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:44:33 +0530
Subject: ASoC: mxs-saif: Handle return value of
 clk_prepare_enable/clk_prepare.

clk_prepare_enable() and clk_prepare() can fail here and
we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mxs/mxs-saif.c | 13 ++++++++++---
 1 file changed, 10 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index b42f301c6b96..156aa7c00787 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -125,7 +125,9 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
 	 *
 	 * If MCLK is not used, we just set saif clk to 512*fs.
 	 */
-	clk_prepare_enable(master_saif->clk);
+	ret = clk_prepare_enable(master_saif->clk);
+	if (ret)
+		return ret;
 
 	if (master_saif->mclk_in_use) {
 		switch (mclk / rate) {
@@ -388,6 +390,7 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *cpu_dai)
 {
 	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret;
 
 	/* clear error status to 0 for each re-open */
 	saif->fifo_underrun = 0;
@@ -401,7 +404,9 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream,
 	__raw_writel(BM_SAIF_CTRL_CLKGATE,
 		saif->base + SAIF_CTRL + MXS_CLR_ADDR);
 
-	clk_prepare(saif->clk);
+	ret = clk_prepare(saif->clk);
+	if (ret)
+		return ret;
 
 	return 0;
 }
@@ -468,7 +473,9 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
 		if (ret)
 			return ret;
 
-		clk_prepare(master_saif->clk);
+		ret = clk_prepare(master_saif->clk);
+		if (ret)
+			return ret;
 	}
 
 	scr = __raw_readl(saif->base + SAIF_CTRL);
-- 
cgit v1.2.3


From eae37345c8653a3c797c1763c95289d924ee37a2 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:44:34 +0530
Subject: ASoC: jz4740: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/jz4740/jz4740-i2s.c | 21 +++++++++++++++++----
 1 file changed, 17 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 794a3499e567..99394c036998 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -133,6 +133,7 @@ static int jz4740_i2s_startup(struct snd_pcm_substream *substream,
 {
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	uint32_t conf, ctrl;
+	int ret;
 
 	if (dai->active)
 		return 0;
@@ -141,7 +142,9 @@ static int jz4740_i2s_startup(struct snd_pcm_substream *substream,
 	ctrl |= JZ_AIC_CTRL_FLUSH;
 	jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
 
-	clk_prepare_enable(i2s->clk_i2s);
+	ret = clk_prepare_enable(i2s->clk_i2s);
+	if (ret)
+		return ret;
 
 	conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
 	conf |= JZ_AIC_CONF_ENABLE;
@@ -352,11 +355,18 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai)
 {
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	uint32_t conf;
+	int ret;
 
-	clk_prepare_enable(i2s->clk_aic);
+	ret = clk_prepare_enable(i2s->clk_aic);
+	if (ret)
+		return ret;
 
 	if (dai->active) {
-		clk_prepare_enable(i2s->clk_i2s);
+		ret = clk_prepare_enable(i2s->clk_i2s);
+		if (ret) {
+			clk_disable_unprepare(i2s->clk_aic);
+			return ret;
+		}
 
 		conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
 		conf |= JZ_AIC_CONF_ENABLE;
@@ -387,8 +397,11 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai)
 {
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	uint32_t conf;
+	int ret;
 
-	clk_prepare_enable(i2s->clk_aic);
+	ret = clk_prepare_enable(i2s->clk_aic);
+	if (ret)
+		return ret;
 
 	jz4740_i2c_init_pcm_config(i2s);
 	snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data,
-- 
cgit v1.2.3


From 3643e9172649f97f2e6301582890202abcc0f97e Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:45:18 +0530
Subject: ASoC: sun4i-spdif: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-spdif.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c
index c49f3757b686..b4af4aabead1 100644
--- a/sound/soc/sunxi/sun4i-spdif.c
+++ b/sound/soc/sunxi/sun4i-spdif.c
@@ -458,11 +458,16 @@ static int sun4i_spdif_runtime_suspend(struct device *dev)
 static int sun4i_spdif_runtime_resume(struct device *dev)
 {
 	struct sun4i_spdif_dev *host  = dev_get_drvdata(dev);
+	int ret;
 
-	clk_prepare_enable(host->spdif_clk);
-	clk_prepare_enable(host->apb_clk);
+	ret = clk_prepare_enable(host->spdif_clk);
+	if (ret)
+		return ret;
+	ret = clk_prepare_enable(host->apb_clk);
+	if (ret)
+		clk_disable_unprepare(host->spdif_clk);
 
-	return 0;
+	return ret;
 }
 
 static int sun4i_spdif_probe(struct platform_device *pdev)
-- 
cgit v1.2.3


From 260ea95cc027500ee5086e2ef568b8c915e66fb5 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Tue, 25 Jul 2017 15:45:37 +0530
Subject: ASoC: atmel: ac97c: Handle return value of clk_prepare_enable.

clk_prepare_enable() can fail here and we must check its return value.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/atmel/ac97c.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 9d2c9d9af688..52b0522fda20 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -783,7 +783,9 @@ static int atmel_ac97c_probe(struct platform_device *pdev)
 		dev_dbg(&pdev->dev, "no peripheral clock\n");
 		return PTR_ERR(pclk);
 	}
-	clk_prepare_enable(pclk);
+	retval = clk_prepare_enable(pclk);
+	if (retval)
+		return retval;
 
 	retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1,
 			      SNDRV_DEFAULT_STR1, THIS_MODULE,
@@ -897,9 +899,9 @@ static int atmel_ac97c_resume(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_ac97c *chip = card->private_data;
+	int ret = clk_prepare_enable(chip->pclk);
 
-	clk_prepare_enable(chip->pclk);
-	return 0;
+	return ret;
 }
 
 static SIMPLE_DEV_PM_OPS(atmel_ac97c_pm, atmel_ac97c_suspend, atmel_ac97c_resume);
-- 
cgit v1.2.3


From 82444cdb4673c146d43f3f5ff3baf582d4f4cb41 Mon Sep 17 00:00:00 2001
From: Pradeep Tewani
Date: Tue, 25 Jul 2017 11:59:46 +0530
Subject: ASoC: Intel: board: Fix missing sentinel for bxt_board_id

Without a sentinel in the array of platform devices, we can get a panic
so add a sentinel. Full stack dump below:

[  231.564705] BUG: KASAN: global-out-of-bounds in platform_match+0xb0/0x146
[  231.565500] Read of size 1 at addr ffffffff8380a1a0 by task swapper/0/1
[  231.566280]
[  231.566594] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.12.0-00289-g152771f #1
[  231.567526] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.9.3-20161025_171302-gandalf 04/01/2014
[  231.568730] Call Trace:
[  231.569128]  dump_stack+0xf8/0x14a
[  231.569615]  print_address_description+0x57/0x1e4
[  231.570216]  ? driver_probe_device+0x814/0x814
[  231.570798]  kasan_report+0x1cb/0x1eb
[  231.571302]  ? platform_match+0xb0/0x146
[  231.571833]  __asan_load1+0x45/0x47
[  231.572324]  platform_match+0xb0/0x146
[  231.572843]  ? platform_uevent+0x73/0x73
[  231.573370]  __driver_attach+0x47/0x16a
[  231.573890]  bus_for_each_dev+0x11a/0x15c
[  231.574422]  ? bus_remove_file+0x57/0x57
[  231.574951]  ? do_raw_spin_unlock+0xc3/0xe0
[  231.575501]  driver_attach+0x2b/0x2e
[  231.576002]  bus_add_driver+0x25e/0x3c5
[  231.576520]  ? set_debug_rodata+0x20/0x20
[  231.584776]  driver_register+0x13e/0x19d
[  231.585305]  __platform_driver_register+0x6c/0x71
[  231.585908]  ? skl_driver_init+0x20/0x20
[  231.586436]  broxton_audio_init+0x17/0x19
[  231.586976]  do_one_initcall+0x11a/0x221
[  231.587499]  ? start_kernel+0x525/0x525
[  231.588020]  ? lock_downgrade+0x281/0x2ec
[  231.588557]  ? trace_hardirqs_on_caller+0x223/0x267
[  231.589169]  ? _raw_spin_unlock_irqrestore+0x49/0x55
[  231.589791]  ? set_debug_rodata+0x20/0x20
[  231.590324]  kernel_init_freeable+0x1fd/0x2b0
[  231.590891]  ? rest_init+0x17c/0x17c
[  231.591384]  kernel_init+0x11/0x157
[  231.591876]  ? rest_init+0x17c/0x17c
[  231.592371]  ret_from_fork+0x2a/0x40
[  231.592865]
[  231.593166] The buggy address belongs to the variable:
[  231.593806]  bxt_board_ids+0x40/0x9e0

Fixes: 152771fbc3 ("ASoC: Intel: board: Add Geminilake platform support")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Pradeep Tewani <pradeep.d.tewani@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/bxt_rt298.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index a4b5bcb1f6f8..7843104fadcb 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -595,6 +595,7 @@ static const struct platform_device_id bxt_board_ids[] = {
 				(unsigned long)&broxton_rt298 },
 	{ .name = "glk_alc298s_i2s", .driver_data =
 				(unsigned long)&geminilake_rt298 },
+	{}
 };
 
 static struct platform_driver broxton_audio = {
-- 
cgit v1.2.3


From ea746a296df3670359ae7d43baa968b0a1a54a01 Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Wed, 26 Jul 2017 20:00:19 +0800
Subject: ASoC: rt5670: merge ADC L/R Mux

Some ADC L/R Muxes share the some control bit. So, merge them.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 42 ++++++++++++++++--------------------------
 1 file changed, 16 insertions(+), 26 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 174141fefc79..0f4e5bc4476f 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -1151,20 +1151,15 @@ static const char * const rt5670_stereo_adc1_src[] = {
 static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER,
 	RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src);
 
-static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux =
-	SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum);
-
-static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux =
-	SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum);
+static const struct snd_kcontrol_new rt5670_sto_adc_1_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC 1 Mux", rt5670_stereo1_adc1_enum);
 
 static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER,
 	RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src);
 
-static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux =
-	SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum);
+static const struct snd_kcontrol_new rt5670_sto2_adc_1_mux =
+	SOC_DAPM_ENUM("Stereo2 ADC 1 Mux", rt5670_stereo2_adc1_enum);
 
-static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux =
-	SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum);
 
 /* MX-27 MX-26 [11] */
 static const char * const rt5670_stereo_adc2_src[] = {
@@ -1174,20 +1169,15 @@ static const char * const rt5670_stereo_adc2_src[] = {
 static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER,
 	RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src);
 
-static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux =
-	SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum);
-
-static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux =
-	SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum);
+static const struct snd_kcontrol_new rt5670_sto_adc_2_mux =
+	SOC_DAPM_ENUM("Stereo1 ADC 2 Mux", rt5670_stereo1_adc2_enum);
 
 static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER,
 	RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src);
 
-static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux =
-	SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum);
+static const struct snd_kcontrol_new rt5670_sto2_adc_2_mux =
+	SOC_DAPM_ENUM("Stereo2 ADC 2 Mux", rt5670_stereo2_adc2_enum);
 
-static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux =
-	SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum);
 
 /* MX-27 MX26 [10] */
 static const char * const rt5670_stereo_adc_src[] = {
@@ -1642,23 +1632,23 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0,
 			 &rt5670_sto1_dmic_mux),
 	SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto_adc_l2_mux),
+			 &rt5670_sto_adc_2_mux),
 	SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto_adc_r2_mux),
+			 &rt5670_sto_adc_2_mux),
 	SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto_adc_l1_mux),
+			 &rt5670_sto_adc_1_mux),
 	SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto_adc_r1_mux),
+			 &rt5670_sto_adc_1_mux),
 	SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0,
 			 &rt5670_sto2_dmic_mux),
 	SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto2_adc_l2_mux),
+			 &rt5670_sto2_adc_2_mux),
 	SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto2_adc_r2_mux),
+			 &rt5670_sto2_adc_2_mux),
 	SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto2_adc_l1_mux),
+			 &rt5670_sto2_adc_1_mux),
 	SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0,
-			 &rt5670_sto2_adc_r1_mux),
+			 &rt5670_sto2_adc_1_mux),
 	SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0,
 			 &rt5670_sto2_adc_lr_mux),
 	SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0,
-- 
cgit v1.2.3


From 776bac64bf13e20c27f7ad168811abdf3b72f14d Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski
Date: Thu, 27 Jul 2017 19:13:36 +0200
Subject: ASoC: samsung: Fix possible double iounmap on s3c24xx driver probe
 failure

Commit 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use
generic dmaengine API") moved ioremap() call from
s3c-i2s-v2.c:s3c_i2sv2_probe() to s3c2412-i2s.c:s3c2412_iis_dev_probe()
and converted it to devm- resource managed interface.

However the error path in first of them - s3c_i2sv2_probe() - was not
updated.  If getting a iis clock in s3c_i2sv2_probe() failed, the
address space would be unmapped there.  This could lead to:
1. double iounmap() later from devm-interface of s3c2412_iis_dev_probe()),
2. accessing the memory by other functions in s3c2412-i2s.c unit.

Anyway, the owner of this mapped region should be s3c2412-i2s.c because
it starts the mapping.

Affected are drivers for S3C24xx family although issue was not reproduced.

Fixes: 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API")
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Arvind Yadav<arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/s3c-i2s-v2.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index ca522a95160b..9b28046eea8e 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -634,7 +634,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai,
 	i2s->iis_pclk = clk_get(dev, "iis");
 	if (IS_ERR(i2s->iis_pclk)) {
 		dev_err(dev, "failed to get iis_clock\n");
-		iounmap(i2s->regs);
 		return -ENOENT;
 	}
 
-- 
cgit v1.2.3


From 81ea6cc73c384e4093519e0c75394aa2a1941a10 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski
Date: Thu, 27 Jul 2017 19:13:37 +0200
Subject: ASoC: samsung: Add missing prepare for iis clock of s3c24xx

The s3c_i2sv2_probe() only enabled iis clock.  Missing prepare isn't
probably fatal, because for SoC clocks this is usually no-op, but for
correctness this clock should be prepared.

Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Arvind Yadav<arvind.yadav.cs@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/s3c-i2s-v2.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 9b28046eea8e..3894bda06ebb 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -637,7 +637,7 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai,
 		return -ENOENT;
 	}
 
-	clk_enable(i2s->iis_pclk);
+	clk_prepare_enable(i2s->iis_pclk);
 
 	/* Mark ourselves as in TXRX mode so we can run through our cleanup
 	 * process without warnings. */
-- 
cgit v1.2.3


From 6f187f7ef1c15dca9f720b18a7e928718fd1a8b3 Mon Sep 17 00:00:00 2001
From: Krzysztof Kozlowski
Date: Thu, 27 Jul 2017 19:13:38 +0200
Subject: ASoC: samsung: Add proper error paths to s3c24xx I2S driver

s3c2412_i2s_probe() might fail so driver has to revert work done by
s3c_i2sv2_probe() (clock enabling).  Missing doing this would lead to
clock enable in-balance.

Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/s3c-i2s-v2.c  |  9 +++++++++
 sound/soc/samsung/s3c-i2s-v2.h  |  7 +++++++
 sound/soc/samsung/s3c2412-i2s.c | 13 +++++++++----
 3 files changed, 25 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 3894bda06ebb..58c3e9bfc6b7 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -651,6 +651,15 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai,
 }
 EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
 
+void s3c_i2sv2_cleanup(struct snd_soc_dai *dai,
+		      struct s3c_i2sv2_info *i2s)
+{
+	clk_disable_unprepare(i2s->iis_pclk);
+	clk_put(i2s->iis_pclk);
+	i2s->iis_pclk = NULL;
+}
+EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup);
+
 #ifdef CONFIG_PM
 static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
 {
diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h
index 182d80564e37..3fca20f7a853 100644
--- a/sound/soc/samsung/s3c-i2s-v2.h
+++ b/sound/soc/samsung/s3c-i2s-v2.h
@@ -91,6 +91,13 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai,
 			   struct s3c_i2sv2_info *i2s,
 			   unsigned long base);
 
+/**
+ * s3c_i2sv2_cleanup - cleanup resources allocated in s3c_i2sv2_probe
+ * @dai: The ASoC DAI structure supplied to the original probe.
+ * @i2s: Our local i2s structure to fill in.
+ */
+extern void s3c_i2sv2_cleanup(struct snd_soc_dai *dai,
+			      struct s3c_i2sv2_info *i2s);
 /**
  * s3c_i2sv2_register_component - register component and dai with soc core
  * @dev: DAI device
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index bcd1cbdeac93..cc0840fff5aa 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -65,7 +65,8 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
 	s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk");
 	if (IS_ERR(s3c2412_i2s.iis_cclk)) {
 		pr_err("failed to get i2sclk clock\n");
-		return PTR_ERR(s3c2412_i2s.iis_cclk);
+		ret = PTR_ERR(s3c2412_i2s.iis_cclk);
+		goto err;
 	}
 
 	/* Set MPLL as the source for IIS CLK */
@@ -73,20 +74,24 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
 	clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
 	ret = clk_prepare_enable(s3c2412_i2s.iis_cclk);
 	if (ret)
-		return ret;
-
-	s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
+		goto err;
 
 	/* Configure the I2S pins (GPE0...GPE4) in correct mode */
 	s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
 			      S3C_GPIO_PULL_NONE);
 
 	return 0;
+
+err:
+	s3c_i2sv2_cleanup(dai, &s3c2412_i2s);
+
+	return ret;
 }
 
 static int s3c2412_i2s_remove(struct snd_soc_dai *dai)
 {
 	clk_disable_unprepare(s3c2412_i2s.iis_cclk);
+	s3c_i2sv2_cleanup(dai, &s3c2412_i2s);
 
 	return 0;
 }
-- 
cgit v1.2.3


From 3c9736aa1ab59e64ebc33b140b000f90c8f9de46 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 27 Jul 2017 01:32:06 +0000
Subject: ASoC: rsnd: merge snd_pcm_ops::open and snd_soc_dai_ops::startup

Current rsnd driver has snd_pcm_ops::open and snd_soc_dai_ops::startup,
but, these are called from soc_pcm_open in same time.
This patch merge these into rsnd_soc_dai_startup()

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/core.c | 80 +++++++++++++++++++-----------------------------
 1 file changed, 31 insertions(+), 49 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index acafb838b7b7..c34562388cfa 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -844,12 +844,28 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
 				ir, &ic);
 }
 
-static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime,
-				   struct snd_soc_dai *dai)
+static struct snd_pcm_hardware rsnd_pcm_hardware = {
+	.info =		SNDRV_PCM_INFO_INTERLEAVED	|
+			SNDRV_PCM_INFO_MMAP		|
+			SNDRV_PCM_INFO_MMAP_VALID,
+	.buffer_bytes_max	= 64 * 1024,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 1,
+	.periods_max		= 32,
+	.fifo_size		= 256,
+};
+
+static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
 {
 	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
 	struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned int max_channels = rsnd_rdai_channels_get(rdai);
+	int ret;
 	int i;
 
 	/*
@@ -866,34 +882,26 @@ static void rsnd_soc_hw_constraint(struct snd_pcm_runtime *runtime,
 		constraint->count = i + 1;
 	}
 
+	snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware);
+
 	snd_pcm_hw_constraint_list(runtime, 0,
 				   SNDRV_PCM_HW_PARAM_CHANNELS, constraint);
 
+	snd_pcm_hw_constraint_integer(runtime,
+				      SNDRV_PCM_HW_PARAM_PERIODS);
+
 	/*
 	 * Sampling Rate / Channel Limitation
 	 * It depends on Clock Master Mode
 	 */
-	if (!rsnd_rdai_is_clk_master(rdai))
-		return;
-
-	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-			    rsnd_soc_hw_rule_rate, dai,
-			    SNDRV_PCM_HW_PARAM_CHANNELS, -1);
-	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			    rsnd_soc_hw_rule_channels, dai,
-			    SNDRV_PCM_HW_PARAM_RATE, -1);
-}
-
-static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
-{
-	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
-	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
-	int ret;
-
-	/* rsnd_io_to_runtime() is not yet enabled here */
-	rsnd_soc_hw_constraint(substream->runtime, dai);
+	if (rsnd_rdai_is_clk_master(rdai)) {
+		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				    rsnd_soc_hw_rule_rate, dai,
+				    SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				    rsnd_soc_hw_rule_channels, dai,
+				    SNDRV_PCM_HW_PARAM_RATE, -1);
+	}
 
 	/*
 	 * call rsnd_dai_call without spinlock
@@ -1105,31 +1113,6 @@ static int rsnd_dai_probe(struct rsnd_priv *priv)
 /*
  *		pcm ops
  */
-static struct snd_pcm_hardware rsnd_pcm_hardware = {
-	.info =		SNDRV_PCM_INFO_INTERLEAVED	|
-			SNDRV_PCM_INFO_MMAP		|
-			SNDRV_PCM_INFO_MMAP_VALID,
-	.buffer_bytes_max	= 64 * 1024,
-	.period_bytes_min	= 32,
-	.period_bytes_max	= 8192,
-	.periods_min		= 1,
-	.periods_max		= 32,
-	.fifo_size		= 256,
-};
-
-static int rsnd_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	int ret = 0;
-
-	snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware);
-
-	ret = snd_pcm_hw_constraint_integer(runtime,
-					    SNDRV_PCM_HW_PARAM_PERIODS);
-
-	return ret;
-}
-
 static int rsnd_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *hw_params)
 {
@@ -1159,7 +1142,6 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream)
 }
 
 static struct snd_pcm_ops rsnd_pcm_ops = {
-	.open		= rsnd_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= rsnd_hw_params,
 	.hw_free	= snd_pcm_lib_free_pages,
-- 
cgit v1.2.3


From 007b6a54c305688d1db7255e5c724e3d76ba5aa4 Mon Sep 17 00:00:00 2001
From: Mihai Serban
Date: Thu, 27 Jul 2017 17:55:10 +0300
Subject: ASoC: codecs: add wm8524 codec driver

WM8524 is a 24-bit 192KHz stereo digital/analog converter (DAC) with
integral charge pump and a simple hardware control interface.
Product information can be found at:
https://www.cirrus.com/products/wm8524/

Signed-off-by: Mihai Serban <mihai.serban@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/wm8524.txt |  16 ++
 sound/soc/codecs/Kconfig                           |   5 +
 sound/soc/codecs/Makefile                          |   2 +
 sound/soc/codecs/wm8524.c                          | 265 +++++++++++++++++++++
 4 files changed, 288 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/wm8524.txt
 create mode 100644 sound/soc/codecs/wm8524.c

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt
new file mode 100644
index 000000000000..20c62002cbcd
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8524.txt
@@ -0,0 +1,16 @@
+WM8524 audio CODEC
+
+This device does not use I2C or SPI but a simple Hardware Control Interface.
+
+Required properties:
+
+  - compatible : "wlf,wm8524"
+
+  - wlf,mute-gpios: a GPIO spec for the MUTE pin.
+
+Example:
+
+codec: wm8524@0 {
+	compatible = "wlf,wm8524";
+	wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>;
+};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 6c78b0b49b81..c3d8aef98608 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -173,6 +173,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8523 if I2C
+	select SND_SOC_WM8524 if GPIOLIB
 	select SND_SOC_WM8580 if I2C
 	select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8727
@@ -967,6 +968,10 @@ config SND_SOC_WM8523
 	tristate "Wolfson Microelectronics WM8523 DAC"
 	depends on I2C
 
+config SND_SOC_WM8524
+	tristate "Wolfson Microelectronics WM8524 DAC"
+	depends on GPIOLIB
+
 config SND_SOC_WM8580
 	tristate "Wolfson Microelectronics WM8580 and WM8581 CODECs"
 	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1755a54e3dc9..66a8567d76d1 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -182,6 +182,7 @@ snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8523-objs := wm8523.o
+snd-soc-wm8524-objs := wm8524.o
 snd-soc-wm8580-objs := wm8580.o
 snd-soc-wm8711-objs := wm8711.o
 snd-soc-wm8727-objs := wm8727.o
@@ -414,6 +415,7 @@ obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
 obj-$(CONFIG_SND_SOC_WM8400)	+= snd-soc-wm8400.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8523)	+= snd-soc-wm8523.o
+obj-$(CONFIG_SND_SOC_WM8524)	+= snd-soc-wm8524.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
 obj-$(CONFIG_SND_SOC_WM8711)	+= snd-soc-wm8711.o
 obj-$(CONFIG_SND_SOC_WM8727)	+= snd-soc-wm8727.o
diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c
new file mode 100644
index 000000000000..f42ec664877c
--- /dev/null
+++ b/sound/soc/codecs/wm8524.c
@@ -0,0 +1,265 @@
+/*
+ * wm8524.c  --  WM8524 ALSA SoC Audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2017 NXP
+ *
+ * Based on WM8523 ALSA SoC Audio driver written by Mark Brown
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/gpio/consumer.h>
+#include <linux/of_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#define WM8524_NUM_RATES 7
+
+/* codec private data */
+struct wm8524_priv {
+	struct gpio_desc *mute;
+	unsigned int sysclk;
+	unsigned int rate_constraint_list[WM8524_NUM_RATES];
+	struct snd_pcm_hw_constraint_list rate_constraint;
+};
+
+
+static const struct snd_soc_dapm_widget wm8524_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("LINEVOUTL"),
+SND_SOC_DAPM_OUTPUT("LINEVOUTR"),
+};
+
+static const struct snd_soc_dapm_route wm8524_dapm_routes[] = {
+	{ "LINEVOUTL", NULL, "DAC" },
+	{ "LINEVOUTR", NULL, "DAC" },
+};
+
+static const struct {
+	int value;
+	int ratio;
+} lrclk_ratios[WM8524_NUM_RATES] = {
+	{ 1, 128 },
+	{ 2, 192 },
+	{ 3, 256 },
+	{ 4, 384 },
+	{ 5, 512 },
+	{ 6, 768 },
+	{ 7, 1152 },
+};
+
+static int wm8524_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec);
+
+	/* The set of sample rates that can be supported depends on the
+	 * MCLK supplied to the CODEC - enforce this.
+	 */
+	if (!wm8524->sysclk) {
+		dev_err(codec->dev,
+			"No MCLK configured, call set_sysclk() on init\n");
+		return -EINVAL;
+	}
+
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_RATE,
+				   &wm8524->rate_constraint);
+
+	gpiod_set_value_cansleep(wm8524->mute, 1);
+
+	return 0;
+}
+
+static void wm8524_shutdown(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec);
+
+	gpiod_set_value_cansleep(wm8524->mute, 0);
+}
+
+static int wm8524_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+	int i, j = 0;
+
+	wm8524->sysclk = freq;
+
+	wm8524->rate_constraint.count = 0;
+	for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+		val = freq / lrclk_ratios[i].ratio;
+		/* Check that it's a standard rate since core can't
+		 * cope with others and having the odd rates confuses
+		 * constraint matching.
+		 */
+		switch (val) {
+		case 8000:
+		case 32000:
+		case 44100:
+		case 48000:
+		case 88200:
+		case 96000:
+		case 176400:
+		case 192000:
+			dev_err(codec->dev, "Supported sample rate: %dHz\n",
+				val);
+			wm8524->rate_constraint_list[j++] = val;
+			wm8524->rate_constraint.count++;
+			break;
+		default:
+			dev_dbg(codec->dev, "Skipping sample rate: %dHz\n",
+				val);
+		}
+	}
+
+	/* Need at least one supported rate... */
+	if (wm8524->rate_constraint.count == 0)
+		return -EINVAL;
+
+	return 0;
+}
+
+static int wm8524_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK |
+		SND_SOC_DAIFMT_MASTER_MASK);
+
+	if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		    SND_SOC_DAIFMT_CBS_CFS)) {
+		dev_err(codec_dai->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
+{
+	struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(dai->codec);
+
+	if (wm8524->mute)
+		gpiod_set_value_cansleep(wm8524->mute, mute);
+
+	return 0;
+}
+
+#define WM8524_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops wm8524_dai_ops = {
+	.startup	= wm8524_startup,
+	.shutdown	= wm8524_shutdown,
+	.set_sysclk	= wm8524_set_dai_sysclk,
+	.set_fmt	= wm8524_set_fmt,
+	.mute_stream	= wm8524_mute_stream,
+};
+
+static struct snd_soc_dai_driver wm8524_dai = {
+	.name = "wm8524-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = WM8524_RATES,
+		.formats = WM8524_FORMATS,
+	},
+	.ops = &wm8524_dai_ops,
+};
+
+static int wm8524_probe(struct snd_soc_codec *codec)
+{
+	struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec);
+
+	wm8524->rate_constraint.list = &wm8524->rate_constraint_list[0];
+	wm8524->rate_constraint.count =
+		ARRAY_SIZE(wm8524->rate_constraint_list);
+
+	return 0;
+}
+
+static const struct snd_soc_codec_driver soc_codec_dev_wm8524 = {
+	.probe =	wm8524_probe,
+
+	.component_driver = {
+		.dapm_widgets		= wm8524_dapm_widgets,
+		.num_dapm_widgets	= ARRAY_SIZE(wm8524_dapm_widgets),
+		.dapm_routes		= wm8524_dapm_routes,
+		.num_dapm_routes	= ARRAY_SIZE(wm8524_dapm_routes),
+	},
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id wm8524_of_match[] = {
+	{ .compatible = "wlf,wm8524" },
+	{ /* sentinel*/ }
+};
+MODULE_DEVICE_TABLE(of, wm8524_of_match);
+#endif
+
+static int wm8524_codec_probe(struct platform_device *pdev)
+{
+	struct wm8524_priv *wm8524;
+	int ret;
+
+	wm8524 = devm_kzalloc(&pdev->dev, sizeof(struct wm8524_priv),
+						  GFP_KERNEL);
+	if (wm8524 == NULL)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, wm8524);
+
+	wm8524->mute = devm_gpiod_get(&pdev->dev, "wlf,mute", GPIOD_OUT_LOW);
+	if (IS_ERR(wm8524->mute)) {
+		ret = PTR_ERR(wm8524->mute);
+		dev_err(&pdev->dev, "Failed to get mute line: %d\n", ret);
+		return ret;
+	}
+
+	ret =  snd_soc_register_codec(&pdev->dev,
+			&soc_codec_dev_wm8524, &wm8524_dai, 1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
+		snd_soc_unregister_platform(&pdev->dev);
+	}
+
+	return ret;
+}
+
+static int wm8524_codec_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver wm8524_codec_driver = {
+	.probe		= wm8524_codec_probe,
+	.remove		= wm8524_codec_remove,
+	.driver		= {
+		.name	= "wm8524-codec",
+		.of_match_table = wm8524_of_match,
+	},
+};
+module_platform_driver(wm8524_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM8524 driver");
+MODULE_AUTHOR("Mihai Serban <mihai.serban@nxp.com>");
+MODULE_ALIAS("platform:wm8524-codec");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From c06425705a2565e6747b0adba6027c7b0b04858f Mon Sep 17 00:00:00 2001
From: Kevin Cheng
Date: Thu, 27 Jul 2017 16:38:36 +0800
Subject: ASoC: Intel: Add Kabylake machine driver for RT5663

This patch adds Kabylake I2S machine driver which uses
RT5663 as headset on ssp1.

Signed-off-by: Kevin Cheng <kevin.cheng@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c | 246 +++++++++++++++++++++++----
 1 file changed, 209 insertions(+), 37 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index f9ba97788157..090331bd6900 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -34,7 +34,7 @@
 #define MAXIM_DEV0_NAME "i2c-MX98927:00"
 #define MAXIM_DEV1_NAME "i2c-MX98927:01"
 
-static struct snd_soc_card kabylake_audio_card;
+static struct snd_soc_card *kabylake_audio_card;
 static const struct snd_pcm_hw_constraint_list *dmic_constraints;
 static struct snd_soc_jack skylake_hdmi[3];
 
@@ -117,6 +117,49 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 	{ "iDisp1 Tx", NULL, "iDisp1_out"},
 };
 
+enum {
+	KBL_DPCM_AUDIO_5663_PB = 0,
+	KBL_DPCM_AUDIO_5663_CP,
+	KBL_DPCM_AUDIO_5663_HDMI1_PB,
+	KBL_DPCM_AUDIO_5663_HDMI2_PB,
+};
+
+static const struct snd_kcontrol_new kabylake_5663_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static const struct snd_soc_dapm_widget kabylake_5663_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_SPK("DP", NULL),
+	SND_SOC_DAPM_SPK("HDMI", NULL),
+};
+
+static const struct snd_soc_dapm_route kabylake_5663_map[] = {
+	{ "Headphone Jack", NULL, "HPOL" },
+	{ "Headphone Jack", NULL, "HPOR" },
+
+	/* other jacks */
+	{ "IN1P", NULL, "Headset Mic" },
+	{ "IN1N", NULL, "Headset Mic" },
+
+	{ "HDMI", NULL, "hif5 Output" },
+	{ "DP", NULL, "hif6 Output" },
+
+	/* CODEC BE connections */
+	{ "AIF Playback", NULL, "ssp1 Tx" },
+	{ "ssp1 Tx", NULL, "codec1_out" },
+
+	{ "codec0_in", NULL, "ssp1 Rx" },
+	{ "ssp1 Rx", NULL, "AIF Capture" },
+
+	{ "hifi2", NULL, "iDisp2 Tx"},
+	{ "iDisp2 Tx", NULL, "iDisp2_out"},
+	{ "hifi1", NULL, "iDisp1 Tx"},
+	{ "iDisp1 Tx", NULL, "iDisp1_out"},
+};
+
 static struct snd_soc_codec_conf max98927_codec_conf[] = {
 	{
 		.dev_name = MAXIM_DEV0_NAME,
@@ -165,7 +208,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
 	 * Headset buttons map to the google Reference headset.
 	 * These can be configured by userspace.
 	 */
-	ret = snd_soc_card_jack_new(&kabylake_audio_card, "Headset Jack",
+	ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack",
 			SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 			SND_JACK_BTN_2 | SND_JACK_BTN_3, &ctx->kabylake_headset,
 			NULL, 0);
@@ -173,8 +216,18 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
 		dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret);
 		return ret;
 	}
-
 	rt5663_set_jack_detect(codec, &ctx->kabylake_headset);
+	return ret;
+}
+
+static int kabylake_rt5663_max98927_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	int ret;
+
+	ret = kabylake_rt5663_codec_init(rtd);
+	if (ret)
+		return ret;
+
 	ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC");
 	if (ret) {
 		dev_err(rtd->dev, "SoC DMIC ignore suspend failed %d\n", ret);
@@ -184,7 +237,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
-static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
+static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
 {
 	struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
@@ -194,7 +247,7 @@ static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
 	if (!pcm)
 		return -ENOMEM;
 
-	pcm->device = KBL_DPCM_AUDIO_HDMI1_PB;
+	pcm->device = device;
 	pcm->codec_dai = dai;
 
 	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
@@ -202,40 +255,29 @@ static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
+static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
-	struct kbl_hdmi_pcm *pcm;
-
-	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
-	if (!pcm)
-		return -ENOMEM;
-
-	pcm->device = KBL_DPCM_AUDIO_HDMI2_PB;
-	pcm->codec_dai = dai;
-
-	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB);
+}
 
-	return 0;
+static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB);
 }
 
 static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
-	struct kbl_hdmi_pcm *pcm;
-
-	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
-	if (!pcm)
-		return -ENOMEM;
-
-	pcm->device = KBL_DPCM_AUDIO_HDMI3_PB;
-	pcm->codec_dai = dai;
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB);
+}
 
-	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+static int kabylake_5663_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_5663_HDMI1_PB);
+}
 
-	return 0;
+static int kabylake_5663_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_5663_HDMI2_PB);
 }
 
 static unsigned int rates[] = {
@@ -538,7 +580,7 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.no_pcm = 1,
 		.codec_name = "i2c-10EC5663:00",
 		.codec_dai_name = KBL_REALTEK_CODEC_DAI,
-		.init = kabylake_rt5663_codec_init,
+		.init = kabylake_rt5663_max98927_codec_init,
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
 		.ignore_pmdown_time = 1,
@@ -594,6 +636,108 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 	},
 };
 
+static struct snd_soc_dai_link kabylake_5663_dais[] = {
+	/* Front End DAI links */
+	[KBL_DPCM_AUDIO_5663_PB] = {
+		.name = "Kbl Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.ops = &kabylake_rt5663_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_5663_CP] = {
+		.name = "Kbl Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+		.ops = &kabylake_rt5663_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_5663_HDMI1_PB] = {
+		.name = "Kbl HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_5663_HDMI2_PB] = {
+		.name = "Kbl HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP1 - Codec */
+		.name = "SSP1-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP1 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codec_name = "i2c-10EC5663:00",
+		.codec_dai_name = KBL_REALTEK_CODEC_DAI,
+		.init = kabylake_rt5663_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = kabylake_ssp_fixup,
+		.ops = &kabylake_rt5663_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "iDisp1",
+		.id = 1,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = kabylake_5663_hdmi1_init,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 2,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_5663_hdmi2_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
 #define NAME_SIZE	32
 static int kabylake_card_late_probe(struct snd_soc_card *card)
 {
@@ -624,7 +768,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card)
 }
 
 /* kabylake audio machine driver for SPT + RT5663 */
-static struct snd_soc_card kabylake_audio_card = {
+static struct snd_soc_card kabylake_audio_card_rt5663_m98927 = {
 	.name = "kblrt5663max",
 	.owner = THIS_MODULE,
 	.dai_link = kabylake_dais,
@@ -641,6 +785,22 @@ static struct snd_soc_card kabylake_audio_card = {
 	.late_probe = kabylake_card_late_probe,
 };
 
+/* kabylake audio machine driver for RT5663 */
+static struct snd_soc_card kabylake_audio_card_rt5663 = {
+	.name = "kblrt5663",
+	.owner = THIS_MODULE,
+	.dai_link = kabylake_5663_dais,
+	.num_links = ARRAY_SIZE(kabylake_5663_dais),
+	.controls = kabylake_5663_controls,
+	.num_controls = ARRAY_SIZE(kabylake_5663_controls),
+	.dapm_widgets = kabylake_5663_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(kabylake_5663_widgets),
+	.dapm_routes = kabylake_5663_map,
+	.num_dapm_routes = ARRAY_SIZE(kabylake_5663_map),
+	.fully_routed = true,
+	.late_probe = kabylake_card_late_probe,
+};
+
 static int kabylake_audio_probe(struct platform_device *pdev)
 {
 	struct kbl_rt5663_private *ctx;
@@ -652,19 +812,30 @@ static int kabylake_audio_probe(struct platform_device *pdev)
 
 	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
 
-	kabylake_audio_card.dev = &pdev->dev;
-	snd_soc_card_set_drvdata(&kabylake_audio_card, ctx);
+	kabylake_audio_card =
+		(struct snd_soc_card *)pdev->id_entry->driver_data;
+
+	kabylake_audio_card->dev = &pdev->dev;
+	snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
 
 	pdata = dev_get_drvdata(&pdev->dev);
 	if (pdata)
 		dmic_constraints = pdata->dmic_num == 2 ?
 			&constraints_dmic_2ch : &constraints_dmic_channels;
 
-	return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card);
+	return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card);
 }
 
 static const struct platform_device_id kbl_board_ids[] = {
-	{ .name = "kbl_rt5663_m98927" },
+	{
+		.name = "kbl_rt5663",
+		.driver_data = (kernel_ulong_t)&kabylake_audio_card_rt5663,
+	},
+	{
+		.name = "kbl_rt5663_m98927",
+		.driver_data =
+			(kernel_ulong_t)&kabylake_audio_card_rt5663_m98927,
+	},
 	{ }
 };
 
@@ -684,4 +855,5 @@ MODULE_DESCRIPTION("Audio Machine driver-RT5663 & MAX98927 in I2S mode");
 MODULE_AUTHOR("Naveen M <naveen.m@intel.com>");
 MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
 MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:kbl_rt5663");
 MODULE_ALIAS("platform:kbl_rt5663_m98927");
-- 
cgit v1.2.3


From 0dfa7a046aa9a3e3dcaee59bfc23598679fc4247 Mon Sep 17 00:00:00 2001
From: Kevin Cheng
Date: Thu, 27 Jul 2017 16:38:37 +0800
Subject: ASoC: Intel: Add Kabylake RT5663 machine driver entry

Adds Kabylake rt5663 machine driver entry into machine table

Signed-off-by: Kevin Cheng <kevin.cheng@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 334917ee41cf..01a2dd6518a4 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -981,6 +981,11 @@ static struct sst_acpi_mach sst_kbl_devdata[] = {
 		.quirk_data = &kbl_poppy_codecs,
 		.pdata = &skl_dmic_data
 	},
+	{
+		.id = "10EC5663",
+		.drv_name = "kbl_rt5663",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+	},
 
 	{}
 };
-- 
cgit v1.2.3


From cc4db0e257d6a5d5ae2e2f042ac8d261caaf08fc Mon Sep 17 00:00:00 2001
From: Harsha Priya N
Date: Thu, 27 Jul 2017 17:41:27 -0700
Subject: ASoC: Intel: Enabling 4 slot IV feedback for max98927 on Kabylake
 platform

This patch enables 4 slot IV feedback for max98927 speaker amp codec.

Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 15 +++++++++++----
 1 file changed, 11 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 3fe4a0807095..74bbc008c965 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -365,11 +365,18 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
 				return ret;
 			}
 		}
-		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME) ||
-			!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
-			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF0, 3, 8, 16);
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
 			if (ret < 0) {
-				dev_err(rtd->dev, "set TDM slot err:%d\n", ret);
+				dev_err(rtd->dev, "DEV0 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+			if (ret < 0) {
+				dev_err(rtd->dev, "DEV1 TDM slot err:%d\n", ret);
 				return ret;
 			}
 		}
-- 
cgit v1.2.3


From 77a4525bc282b7072ebda5d6f6925ca6d604a16f Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Wed, 26 Jul 2017 02:35:07 +0200
Subject: ASoC: jack: fix snd_soc_codec_set_jack return error

This patch changes the error code returned by snd_soc_codec_set_jack()
from -EINVAL to -ENOTSUPP. The reason to do this is to make the caller
aware that the underlying codec does not support this callback. This can
make the caller write the code to handle this case properly.
Other reason is that -EINVAL is not the correct error to return in this
case anyway.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-jack.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7daf21fee355..42ca9f19843b 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -36,7 +36,7 @@ int snd_soc_codec_set_jack(struct snd_soc_codec *codec,
 	if (codec->driver->set_jack)
 		return codec->driver->set_jack(codec, jack, data);
 	else
-		return -EINVAL;
+		return -ENOTSUPP;
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack);
 
-- 
cgit v1.2.3


From f1e3f409d61372b742697d0b5e5cd17d09de37db Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Sat, 29 Jul 2017 11:40:55 -0300
Subject: ASoC: soc-pcm: Remove unused 'debugfs_dpcm_state' entry

'debugfs_dpcm_state' member from structure snd_soc_pcm_runtime
is never used at all, so it is safe to remove it.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h | 1 -
 sound/soc/soc-pcm.c | 5 ++---
 2 files changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9c94b97c17f8..230b8eedd9f2 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1240,7 +1240,6 @@ struct snd_soc_pcm_runtime {
 	struct delayed_work delayed_work;
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_dpcm_root;
-	struct dentry *debugfs_dpcm_state;
 #endif
 
 	unsigned int num; /* 0-based and monotonic increasing */
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index dcc5ece08668..2ddf6aae230d 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -3010,8 +3010,7 @@ void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
 		return;
 	}
 
-	rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444,
-						rtd->debugfs_dpcm_root,
-						rtd, &dpcm_state_fops);
+	debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root,
+			    rtd, &dpcm_state_fops);
 }
 #endif
-- 
cgit v1.2.3


From bf11ecb7eebd28d70bf5aa3170c6fe7156c33c5b Mon Sep 17 00:00:00 2001
From: Arnd Bergmann
Date: Mon, 31 Jul 2017 18:05:46 +0200
Subject: ASoC: codecs: fix wm8524 build error

When CONFIG_OF is disabled, we run into a build error:

sound/soc/codecs/wm8524.c:257:21: error: 'wm8524_of_match' undeclared here (not in a function); did you mean 'wm8524_dai'?

This removes the unnecessary #ifdef around the match table.

Fixes: 007b6a54c305 ("ASoC: codecs: add wm8524 codec driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Mihai Serban <mihai.serban@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8524.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c
index f42ec664877c..e22b76d64e32 100644
--- a/sound/soc/codecs/wm8524.c
+++ b/sound/soc/codecs/wm8524.c
@@ -206,13 +206,11 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8524 = {
 	},
 };
 
-#ifdef CONFIG_OF
 static const struct of_device_id wm8524_of_match[] = {
 	{ .compatible = "wlf,wm8524" },
 	{ /* sentinel*/ }
 };
 MODULE_DEVICE_TABLE(of, wm8524_of_match);
-#endif
 
 static int wm8524_codec_probe(struct platform_device *pdev)
 {
-- 
cgit v1.2.3


From c1bbaff6d7dd55314ee453df8dfe00befd7ce1b2 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Tue, 1 Aug 2017 19:14:02 +0800
Subject: ASoC: rt5663: Add the delay time to correct the calibration

The patch adds the delay time to correct the calibration.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 45f8a9888f7d..c1f0a72c0133 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -3058,6 +3058,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663)
 	regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c);
 	regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32);
 	regmap_write(rt5663->regmap, RT5663_DACREF_LDO, 0x3b0b);
+	msleep(40);
 	regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x0000);
 	regmap_write(rt5663->regmap, RT5663_BYPASS_STO_DAC, 0x000c);
 	regmap_write(rt5663->regmap, RT5663_HP_BIAS, 0xafaa);
-- 
cgit v1.2.3


From 60767abcea3dd1d47a4bd38398e1e8c1864975eb Mon Sep 17 00:00:00 2001
From: guneshwor.o.singh@intel.com
Date: Fri, 28 Jul 2017 16:12:13 +0530
Subject: ASoC: Intel: Skylake: Reset the controller in probe

Controller can be in reset state by default. Capability structure
traversal requires the controller to be out of reset else it
results in broken capability parsing. Hence make sure that controller is
out of reset before parsing capabilities by doing a full reset.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 01a2dd6518a4..b9e1310673f0 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -702,6 +702,8 @@ static int skl_first_init(struct hdac_ext_bus *ebus)
 		return -ENXIO;
 	}
 
+	skl_init_chip(bus, true);
+
 	snd_hdac_bus_parse_capabilities(bus);
 
 	if (skl_acquire_irq(ebus, 0) < 0)
-- 
cgit v1.2.3


From 3fbbcf4dadb6c01d6411bac715a7448f96c3724c Mon Sep 17 00:00:00 2001
From: Dronamraju Santosh P K
Date: Fri, 28 Jul 2017 16:12:14 +0530
Subject: ASoC: Intel: bxtn: Remove code loader reference in cleanup

Since Broxton does not use code loader DMA, remove code loader cleanup
in its dsp cleanup routine. Remove the iounmap too as it is done in
skl_free_dsp().

Signed-off-by: Dronamraju Santosh P K <santosh.pavan.kumarx.dronamraju@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/bxt-sst.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index cf11b84888b9..08a2c5eb59fa 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -629,11 +629,6 @@ void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
 		release_firmware(ctx->dsp->fw);
 	skl_freeup_uuid_list(ctx);
 	skl_ipc_free(&ctx->ipc);
-	ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
-
-	if (ctx->dsp->addr.lpe)
-		iounmap(ctx->dsp->addr.lpe);
-
 	ctx->dsp->ops->free(ctx->dsp);
 }
 EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup);
-- 
cgit v1.2.3


From 693c0fb26a312d7cc866b1e9195187789b69e7e6 Mon Sep 17 00:00:00 2001
From: guneshwor.o.singh@intel.com
Date: Fri, 28 Jul 2017 16:12:15 +0530
Subject: ASoC: Intel: Skylake: Remove return check for skl_codec_create()

Since skl_codec_create() always returns 0, make it return void and remove
return check.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl.c | 8 ++------
 1 file changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index b9e1310673f0..ab651517e8b8 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -528,7 +528,7 @@ static int probe_codec(struct hdac_ext_bus *ebus, int addr)
 }
 
 /* Codec initialization */
-static int skl_codec_create(struct hdac_ext_bus *ebus)
+static void skl_codec_create(struct hdac_ext_bus *ebus)
 {
 	struct hdac_bus *bus = ebus_to_hbus(ebus);
 	int c, max_slots;
@@ -559,8 +559,6 @@ static int skl_codec_create(struct hdac_ext_bus *ebus)
 			}
 		}
 	}
-
-	return 0;
 }
 
 static const struct hdac_bus_ops bus_core_ops = {
@@ -612,9 +610,7 @@ static void skl_probe_work(struct work_struct *work)
 		dev_info(bus->dev, "no hda codecs found!\n");
 
 	/* create codec instances */
-	err = skl_codec_create(ebus);
-	if (err < 0)
-		goto out_err;
+	skl_codec_create(ebus);
 
 	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
 		err = snd_hdac_display_power(bus, false);
-- 
cgit v1.2.3


From 10a5439f0e4db0b1c4c2a2f6c916d6501cfeae50 Mon Sep 17 00:00:00 2001
From: guneshwor.o.singh@intel.com
Date: Fri, 28 Jul 2017 16:12:16 +0530
Subject: ASoC: Intel: Skylake: Fix potential null pointer dereference

Check if the next sink is not null to avoid potential null pointer
dereference in skl_tplg_bind_sinks().

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 68c3f121efc3..8c3ce08f366d 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -925,7 +925,7 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
 		}
 	}
 
-	if (!sink)
+	if (!sink && next_sink)
 		return skl_tplg_bind_sinks(next_sink, skl, src_w, src_mconfig);
 
 	return 0;
-- 
cgit v1.2.3


From 47bea0c836867b6b1cdb714d58f2bfc8e2f5c386 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 29 Jul 2017 16:17:42 +0200
Subject: ASoC: sun4i-i2s: Extend quirks scope

In preparation for the changes required to support newer SoCs then
quirks has been moved and also added to the device structure.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 22 ++++++++++++++--------
 1 file changed, 14 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 62b307b0c846..d7ee7a443e4e 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -90,6 +90,15 @@
 #define SUN4I_I2S_RX_CHAN_SEL_REG	0x38
 #define SUN4I_I2S_RX_CHAN_MAP_REG	0x3c
 
+/**
+ * struct sun4i_i2s_quirks - Differences between SoC variants.
+ *
+ * @has_reset: SoC needs reset deasserted.
+ */
+struct sun4i_i2s_quirks {
+	bool				has_reset;
+};
+
 struct sun4i_i2s {
 	struct clk	*bus_clk;
 	struct clk	*mod_clk;
@@ -100,6 +109,8 @@ struct sun4i_i2s {
 
 	struct snd_dmaengine_dai_dma_data	capture_dma_data;
 	struct snd_dmaengine_dai_dma_data	playback_dma_data;
+
+	const struct sun4i_i2s_quirks	*variant;
 };
 
 struct sun4i_i2s_clk_div {
@@ -654,10 +665,6 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
 	return 0;
 }
 
-struct sun4i_i2s_quirks {
-	bool has_reset;
-};
-
 static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.has_reset	= false,
 };
@@ -669,7 +676,6 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 static int sun4i_i2s_probe(struct platform_device *pdev)
 {
 	struct sun4i_i2s *i2s;
-	const struct sun4i_i2s_quirks *quirks;
 	struct resource *res;
 	void __iomem *regs;
 	int irq, ret;
@@ -690,8 +696,8 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 		return irq;
 	}
 
-	quirks = of_device_get_match_data(&pdev->dev);
-	if (!quirks) {
+	i2s->variant = of_device_get_match_data(&pdev->dev);
+	if (!i2s->variant) {
 		dev_err(&pdev->dev, "Failed to determine the quirks to use\n");
 		return -ENODEV;
 	}
@@ -715,7 +721,7 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 		return PTR_ERR(i2s->mod_clk);
 	}
 
-	if (quirks->has_reset) {
+	if (i2s->variant->has_reset) {
 		i2s->rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
 		if (IS_ERR(i2s->rst)) {
 			dev_err(&pdev->dev, "Failed to get reset control\n");
-- 
cgit v1.2.3


From 818010dac09183e99ff02e35836544b2e1d659f3 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Mon, 31 Jul 2017 13:47:42 +0800
Subject: ASoC: rt5514: Add the sanity checks of the buffer related address

The patch add the sanity checks of the buffer related address to make sure
the addresses are valid.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514-spi.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 950d1ffdc06c..6f513cc4c56c 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -173,9 +173,14 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data)
 	rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 |
 				buf[3] << 24;
 
+	if (rt5514_dsp->buf_rp % 8)
+		rt5514_dsp->buf_rp = (rt5514_dsp->buf_rp / 8) * 8;
+
 	rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base;
 
-	schedule_delayed_work(&rt5514_dsp->copy_work, 0);
+	if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit &&
+		rt5514_dsp->buf_rp && rt5514_dsp->buf_size)
+		schedule_delayed_work(&rt5514_dsp->copy_work, 0);
 
 	return IRQ_HANDLED;
 }
-- 
cgit v1.2.3


From 19b4b7292615358b9599ff9410a3e564b4844e90 Mon Sep 17 00:00:00 2001
From: Axel Lin
Date: Wed, 2 Aug 2017 17:55:13 +0800
Subject: ASoC: wm8523: Fix array size for bclk_ratios

ARRAY_SIZE(bclk_ratios) returns 7 for current code, then it cannot catch
the error if "no matching BCLK/fs ratio". Fix it.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8523.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 6d0a2723bfde..b1e4ce72e533 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -116,7 +116,7 @@ static struct {
 static struct {
 	int value;
 	int ratio;
-} bclk_ratios[WM8523_NUM_RATES] = {
+} bclk_ratios[] = {
 	{ 2, 32 },
 	{ 3, 64 },
 	{ 4, 128 },
-- 
cgit v1.2.3


From c0d088f04f115409287003b5fc0dfd3184d7d306 Mon Sep 17 00:00:00 2001
From: Axel Lin
Date: Wed, 2 Aug 2017 17:55:14 +0800
Subject: ASoC: wm8523: Constfiy lrclk_ratios and bclk_ratios

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8523.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index b1e4ce72e533..c7c33e98fbcb 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -100,7 +100,7 @@ static const struct snd_soc_dapm_route wm8523_dapm_routes[] = {
 	{ "LINEVOUTR", NULL, "DAC" },
 };
 
-static struct {
+static const struct {
 	int value;
 	int ratio;
 } lrclk_ratios[WM8523_NUM_RATES] = {
@@ -113,7 +113,7 @@ static struct {
 	{ 7, 1152 },
 };
 
-static struct {
+static const struct {
 	int value;
 	int ratio;
 } bclk_ratios[] = {
-- 
cgit v1.2.3


From 80bfd0431f7dc133f4edbc0b864f800c4ed7a3f4 Mon Sep 17 00:00:00 2001
From: Axel Lin
Date: Wed, 2 Aug 2017 16:13:46 +0800
Subject: ASoC: wm8524: Don't use dev_err to show supported sample rate

Switch to use dev_dbg instead.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Mihai Serban <mihai.serban@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8524.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c
index e22b76d64e32..b66147e0631b 100644
--- a/sound/soc/codecs/wm8524.c
+++ b/sound/soc/codecs/wm8524.c
@@ -118,7 +118,7 @@ static int wm8524_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		case 96000:
 		case 176400:
 		case 192000:
-			dev_err(codec->dev, "Supported sample rate: %dHz\n",
+			dev_dbg(codec->dev, "Supported sample rate: %dHz\n",
 				val);
 			wm8524->rate_constraint_list[j++] = val;
 			wm8524->rate_constraint.count++;
-- 
cgit v1.2.3


From 278982b50b28d654ff8cca0efb1ff6e835c6cc4c Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Wed, 2 Aug 2017 16:01:27 +0800
Subject: ASoC: rt5663: Seprate the DC offset between headphone and headset

The patch seprates the DC offset between headphone and headset.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/rt5663.txt |  5 ++-
 include/sound/rt5663.h                             |  2 +
 sound/soc/codecs/rt5663.c                          | 52 ++++++++++++++++------
 3 files changed, 44 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt
index f7c7cc513ee8..ff381718c517 100644
--- a/Documentation/devicetree/bindings/sound/rt5663.txt
+++ b/Documentation/devicetree/bindings/sound/rt5663.txt
@@ -14,8 +14,11 @@ Optional properties:
 
 - "realtek,dc_offset_l_manual"
 - "realtek,dc_offset_r_manual"
+- "realtek,dc_offset_l_manual_mic"
+- "realtek,dc_offset_r_manual_mic"
   Based on the different PCB layout, add the manual offset value to
-  compensate the DC offset for each L and R channel.
+  compensate the DC offset for each L and R channel, and they are different
+  between headphone and headset.
 
 Pins on the device (for linking into audio routes) for RT5663:
 
diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h
index be6f1b2d8fbd..7d00e5849706 100644
--- a/include/sound/rt5663.h
+++ b/include/sound/rt5663.h
@@ -14,6 +14,8 @@
 struct rt5663_platform_data {
 	unsigned int dc_offset_l_manual;
 	unsigned int dc_offset_r_manual;
+	unsigned int dc_offset_l_manual_mic;
+	unsigned int dc_offset_r_manual_mic;
 };
 
 #endif
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index c1f0a72c0133..be93231b0f58 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -1570,9 +1570,43 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 		case 2:
 			rt5663->jack_type = SND_JACK_HEADSET;
 			rt5663_enable_push_button_irq(codec, true);
+
+			if (rt5663->pdata.dc_offset_l_manual_mic) {
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2,
+					rt5663->pdata.dc_offset_l_manual_mic >>
+					16);
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3,
+					rt5663->pdata.dc_offset_l_manual_mic &
+					0xffff);
+			}
+
+			if (rt5663->pdata.dc_offset_r_manual_mic) {
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5,
+					rt5663->pdata.dc_offset_r_manual_mic >>
+					16);
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6,
+					rt5663->pdata.dc_offset_r_manual_mic &
+					0xffff);
+			}
 			break;
 		default:
 			rt5663->jack_type = SND_JACK_HEADPHONE;
+
+			if (rt5663->pdata.dc_offset_l_manual) {
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2,
+					rt5663->pdata.dc_offset_l_manual >> 16);
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3,
+					rt5663->pdata.dc_offset_l_manual &
+					0xffff);
+			}
+
+			if (rt5663->pdata.dc_offset_r_manual) {
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5,
+					rt5663->pdata.dc_offset_r_manual >> 16);
+				regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6,
+					rt5663->pdata.dc_offset_r_manual &
+					0xffff);
+			}
 			break;
 		}
 	} else {
@@ -3133,6 +3167,10 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev)
 		&rt5663->pdata.dc_offset_l_manual);
 	device_property_read_u32(dev, "realtek,dc_offset_r_manual",
 		&rt5663->pdata.dc_offset_r_manual);
+	device_property_read_u32(dev, "realtek,dc_offset_l_manual_mic",
+		&rt5663->pdata.dc_offset_l_manual_mic);
+	device_property_read_u32(dev, "realtek,dc_offset_r_manual_mic",
+		&rt5663->pdata.dc_offset_r_manual_mic);
 
 	return 0;
 }
@@ -3221,20 +3259,6 @@ static int rt5663_i2c_probe(struct i2c_client *i2c,
 		dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__);
 	}
 
-	if (rt5663->pdata.dc_offset_l_manual) {
-		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2,
-			rt5663->pdata.dc_offset_l_manual >> 16);
-		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3,
-			rt5663->pdata.dc_offset_l_manual & 0xffff);
-	}
-
-	if (rt5663->pdata.dc_offset_r_manual) {
-		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5,
-			rt5663->pdata.dc_offset_r_manual >> 16);
-		regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6,
-			rt5663->pdata.dc_offset_r_manual & 0xffff);
-	}
-
 	/* GPIO1 as IRQ */
 	regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5663_GP1_PIN_MASK,
 		RT5663_GP1_PIN_IRQ);
-- 
cgit v1.2.3


From 29685e207b37ad7f415e4bf21320076c025f3df1 Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen
Date: Tue, 25 Jul 2017 10:48:14 +0200
Subject: ASoC: codec: add DT support in dmic codec

Add of_table to allows DT probing.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/dmic.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index c82b9dc41e9a..d50f142e2972 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -73,9 +73,15 @@ static int dmic_dev_remove(struct platform_device *pdev)
 
 MODULE_ALIAS("platform:dmic-codec");
 
+static const struct of_device_id dmic_dev_match[] = {
+	{.compatible = "dmic-codec"},
+	{}
+};
+
 static struct platform_driver dmic_driver = {
 	.driver = {
 		.name = "dmic-codec",
+		.of_match_table = dmic_dev_match,
 	},
 	.probe = dmic_dev_probe,
 	.remove = dmic_dev_remove,
-- 
cgit v1.2.3


From e3c6de48133ff1355f14a31a74a9027834c0fd65 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Wed, 2 Aug 2017 10:26:09 +0000
Subject: ASoC: rcar: unregister fixed rate on ADG

ADG is registering fixed rate clock for audio_clkout, but it had not
been unregister clock when removing.

Salvator-X board is using ak4613 driver now, and it supports
hw_constraints from commit 907cd8809eebc ("ASoC: ak4613: add
hw_constraint rule for Sampling Rate").
And this calculation is using input clk. This ak4613 input clock is
ADG clkout on Salvator-X.

Because ADG had not been unregister clkout when unbinding, it receives
fixed rate clk register error when re-binding.
Thus, ak4613 can't get correct input clock, and hw_constraints will be
failed after re-binding.
This means Salvator-X board can't use sound after unbind/bind.

This patch solves this issue.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/adg.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 197cb3ec075f..5b5389e5b92b 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -610,6 +610,13 @@ void rsnd_adg_remove(struct rsnd_priv *priv)
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct device_node *np = dev->of_node;
+	struct rsnd_adg *adg = priv->adg;
+	struct clk *clk;
+	int i;
+
+	for_each_rsnd_clkout(clk, adg, i)
+		if (adg->clkout[i])
+			clk_unregister_fixed_rate(adg->clkout[i]);
 
 	of_clk_del_provider(np);
 
-- 
cgit v1.2.3


From 2032ce4de818366adb78d8e0b29291ce58ae1e40 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Wed, 2 Aug 2017 15:17:46 +0200
Subject: ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk

This patch adds support to set_sysclk() which can let the sound
card driver to set default mclk rate. In this case MCLK for
internal audio codec is expected to be at 9.6MHz by default.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-digital.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 7e3794fb8c2c..661cd6dd5473 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -568,6 +568,15 @@ static int msm8916_wcd_digital_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static int msm8916_wcd_digital_codec_set_sysclk(struct snd_soc_codec *codec,
+						int clk_id, int source,
+						unsigned int freq, int dir)
+{
+	struct msm8916_wcd_digital_priv *p = dev_get_drvdata(codec->dev);
+
+	return clk_set_rate(p->mclk, freq);
+}
+
 static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream,
 					 struct snd_pcm_hw_params *params,
 					 struct snd_soc_dai *dai)
@@ -823,6 +832,7 @@ static struct snd_soc_dai_driver msm8916_wcd_digital_dai[] = {
 
 static struct snd_soc_codec_driver msm8916_wcd_digital = {
 	.probe = msm8916_wcd_digital_codec_probe,
+	.set_sysclk = msm8916_wcd_digital_codec_set_sysclk,
 	.component_driver = {
 		.controls = msm8916_wcd_digital_snd_controls,
 		.num_controls = ARRAY_SIZE(msm8916_wcd_digital_snd_controls),
-- 
cgit v1.2.3


From 820f7541f952afa8db08ccd420f29a2c6c148709 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Wed, 2 Aug 2017 15:17:46 +0200
Subject: ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk

This patch adds support to set_sysclk() which can let the sound
card driver to set default mclk rate. In this case MCLK for
internal audio codec is expected to be at 9.6MHz by default.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-digital.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index f690442af8c9..825cc7d70bb3 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -568,6 +568,15 @@ static int msm8916_wcd_digital_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static int msm8916_wcd_digital_codec_set_sysclk(struct snd_soc_codec *codec,
+						int clk_id, int source,
+						unsigned int freq, int dir)
+{
+	struct msm8916_wcd_digital_priv *p = dev_get_drvdata(codec->dev);
+
+	return clk_set_rate(p->mclk, freq);
+}
+
 static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream,
 					 struct snd_pcm_hw_params *params,
 					 struct snd_soc_dai *dai)
@@ -823,6 +832,7 @@ static struct snd_soc_dai_driver msm8916_wcd_digital_dai[] = {
 
 static struct snd_soc_codec_driver msm8916_wcd_digital = {
 	.probe = msm8916_wcd_digital_codec_probe,
+	.set_sysclk = msm8916_wcd_digital_codec_set_sysclk,
 	.component_driver = {
 		.controls = msm8916_wcd_digital_snd_controls,
 		.num_controls = ARRAY_SIZE(msm8916_wcd_digital_snd_controls),
-- 
cgit v1.2.3


From 334822a3b21f5b622ae1954915003d636186e7b0 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Wed, 2 Aug 2017 15:17:47 +0200
Subject: ASoC: qcom: apq8016-sbc: set default mclk rate

MCLK for internal audio codec is expected to be at 9.6MHz by default.
This patch adds support to 9.6MHz to make the default case possible.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/apq8016_sbc.c | 23 +++++++++++++++++++----
 1 file changed, 19 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index d084d7468299..f07aa1e1cdfe 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -34,13 +34,16 @@ struct apq8016_sbc_data {
 #define MIC_CTRL_QUA_WS_SLAVE_SEL_10	BIT(17)
 #define MIC_CTRL_TLMM_SCLK_EN		BIT(1)
 #define	SPKR_CTL_PRI_WS_SLAVE_SEL_11	(BIT(17) | BIT(16))
+#define DEFAULT_MCLK_RATE		9600000
 
 static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_codec *codec;
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 	struct snd_soc_card *card = rtd->card;
 	struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card);
-	int rval = 0;
+	int i, rval;
 
 	switch (cpu_dai->id) {
 	case MI2S_PRIMARY:
@@ -63,12 +66,24 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
 
 	default:
 		dev_err(card->dev, "unsupported cpu dai configuration\n");
-		rval = -EINVAL;
-		break;
+		return -EINVAL;
+
+	}
 
+	for (i = 0 ; i < dai_link->num_codecs; i++) {
+		struct snd_soc_dai *dai = rtd->codec_dais[i];
+
+		codec = dai->codec;
+		/* Set default mclk for internal codec */
+		rval = snd_soc_codec_set_sysclk(codec, 0, 0, DEFAULT_MCLK_RATE,
+				       SND_SOC_CLOCK_IN);
+		if (rval != 0 && rval != -ENOTSUPP) {
+			dev_warn(card->dev, "Failed to set mclk: %d\n", rval);
+			return rval;
+		}
 	}
 
-	return rval;
+	return 0;
 }
 
 static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
-- 
cgit v1.2.3


From 3a88a3757dc654c29c87f4537053d0f3dc28d9e0 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Wed, 2 Aug 2017 15:17:48 +0200
Subject: ASoC: codecs: msm8916-wcd-digital: add CIC filter source selection
 path

This patch fixes a missing selection of DMIC in CIC filter source path
to dapm route. Without this patch dmic is not functional.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-digital.c | 25 +++++++++++++++++++++++--
 1 file changed, 23 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 825cc7d70bb3..d2993e72d1e2 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -218,6 +218,8 @@ static const char *const rx_mix1_text[] = {
 static const char *const dec_mux_text[] = {
 	"ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2"
 };
+
+static const char *const cic_mux_text[] = { "AMIC", "DMIC" };
 static const char *const rx_mix2_text[] = { "ZERO", "IIR1", "IIR2" };
 static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
 
@@ -256,11 +258,21 @@ static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE(
 static const struct soc_enum dec2_mux_enum = SOC_ENUM_SINGLE(
 				LPASS_CDC_CONN_TX_B1_CTL, 3, 6, dec_mux_text);
 
+/* CIC */
+static const struct soc_enum cic1_mux_enum = SOC_ENUM_SINGLE(
+				LPASS_CDC_TX1_MUX_CTL, 0, 2, cic_mux_text);
+static const struct soc_enum cic2_mux_enum = SOC_ENUM_SINGLE(
+				LPASS_CDC_TX2_MUX_CTL, 0, 2, cic_mux_text);
+
 /* RDAC2 MUX */
 static const struct snd_kcontrol_new dec1_mux = SOC_DAPM_ENUM(
 				"DEC1 MUX Mux", dec1_mux_enum);
 static const struct snd_kcontrol_new dec2_mux = SOC_DAPM_ENUM(
 				"DEC2 MUX Mux",	dec2_mux_enum);
+static const struct snd_kcontrol_new cic1_mux = SOC_DAPM_ENUM(
+				"CIC1 MUX Mux", cic1_mux_enum);
+static const struct snd_kcontrol_new cic2_mux = SOC_DAPM_ENUM(
+				"CIC2 MUX Mux",	cic2_mux_enum);
 static const struct snd_kcontrol_new rx_mix1_inp1_mux = SOC_DAPM_ENUM(
 				"RX1 MIX1 INP1 Mux", rx_mix1_inp_enum[0]);
 static const struct snd_kcontrol_new rx_mix1_inp2_mux = SOC_DAPM_ENUM(
@@ -500,6 +512,8 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0,
 			 &rx3_mix1_inp3_mux),
 
+	SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux),
+	SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux),
 	/* TX */
 	SND_SOC_DAPM_MIXER("ADC1", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_MIXER("ADC2", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -536,6 +550,8 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = {
 	/* Connectivity Clock */
 	SND_SOC_DAPM_SUPPLY_S("CDC_CONN", -2, LPASS_CDC_CLK_OTHR_CTL, 2, 0,
 			      NULL, 0),
+	SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+	SND_SOC_DAPM_MIC("Digital Mic2", NULL),
 
 };
 
@@ -655,6 +671,11 @@ static const struct snd_soc_dapm_route msm8916_wcd_digital_audio_map[] = {
 	{"AIF1 Capture", NULL, "I2S TX2"},
 	{"AIF1 Capture", NULL, "I2S TX3"},
 
+	{"CIC1 MUX", "DMIC", "DEC1 MUX"},
+	{"CIC1 MUX", "AMIC", "DEC1 MUX"},
+	{"CIC2 MUX", "DMIC", "DEC2 MUX"},
+	{"CIC2 MUX", "AMIC", "DEC2 MUX"},
+
 	/* Decimator Inputs */
 	{"DEC1 MUX", "DMIC1", "DMIC1"},
 	{"DEC1 MUX", "DMIC2", "DMIC2"},
@@ -673,8 +694,8 @@ static const struct snd_soc_dapm_route msm8916_wcd_digital_audio_map[] = {
 	{"DMIC1", NULL, "DMIC_CLK"},
 	{"DMIC2", NULL, "DMIC_CLK"},
 
-	{"I2S TX1", NULL, "DEC1 MUX"},
-	{"I2S TX2", NULL, "DEC2 MUX"},
+	{"I2S TX1", NULL, "CIC1 MUX"},
+	{"I2S TX2", NULL, "CIC2 MUX"},
 
 	{"I2S TX1", NULL, "TX_I2S_CLK"},
 	{"I2S TX2", NULL, "TX_I2S_CLK"},
-- 
cgit v1.2.3


From d70c4a048195769ff22e65b9efe984de2259b974 Mon Sep 17 00:00:00 2001
From: Naveen Manohar
Date: Wed, 2 Aug 2017 11:47:43 +0530
Subject: ASoC: Intel: Skylake: Use correct nuvoton codec ID

Correcting the nau88l25 codec ID to match secondary codec
and load the machine driver.

Signed-off-by: Naveen M <naveen.m@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 9e3f8c04dd32..6b49feb11b12 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -879,12 +879,12 @@ static void skl_remove(struct pci_dev *pci)
 
 static struct sst_codecs skl_codecs = {
 	.num_codecs = 1,
-	.codecs = {"NAU88L25"}
+	.codecs = {"10508825"}
 };
 
 static struct sst_codecs kbl_codecs = {
 	.num_codecs = 1,
-	.codecs = {"NAU88L25"}
+	.codecs = {"10508825"}
 };
 
 static struct sst_codecs bxt_codecs = {
-- 
cgit v1.2.3


From da3cbb40f35b4b5ab66df9a23a717a4a5df1f992 Mon Sep 17 00:00:00 2001
From: Naveen Manohar
Date: Wed, 2 Aug 2017 11:43:33 +0530
Subject: ASoC: Intel: kbl: Add new FEs for Multi-Playback & Echo-Reference

Adding new frontends to support:
1. Parallel playback on 2 ports simultaneously
2. Echo reference capture capability

Signed-off-by: Naveen M <naveen.m@intel.com>
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-pcm.c | 26 ++++++++++++++++++++++++++
 1 file changed, 26 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index b1f2dba642c4..debdaac2c3a2 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -674,6 +674,32 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 		.sig_bits = 32,
 	},
 },
+{
+	.name = "System Pin2",
+	.ops = &skl_pcm_dai_ops,
+	.playback = {
+		.stream_name = "Headset Playback",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000 |
+			SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
+	},
+},
+{
+	.name = "Echoref Pin",
+	.ops = &skl_pcm_dai_ops,
+	.capture = {
+		.stream_name = "Echoreference Capture",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000 |
+			SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
+	},
+},
 {
 	.name = "Reference Pin",
 	.ops = &skl_pcm_dai_ops,
-- 
cgit v1.2.3


From b32ee384ac18d3e4db0ef45d23c3ff1c67113529 Mon Sep 17 00:00:00 2001
From: Naveen Manohar
Date: Wed, 2 Aug 2017 11:43:34 +0530
Subject: ASoC: Intel: kbl: Add DAI links for Multi-Playback & Echo-reference

Add new FE DAI links to support:
1. Parallel playback on 2 ports simultaneously
2. Echo reference capture capability

Signed-off-by: Naveen M <naveen.m@intel.com>
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c       | 24 ++++++++++++++++++++++
 .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c  | 24 ++++++++++++++++++++++
 2 files changed, 48 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 090331bd6900..4c70d5a3028c 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -52,6 +52,8 @@ struct kbl_rt5663_private {
 enum {
 	KBL_DPCM_AUDIO_PB = 0,
 	KBL_DPCM_AUDIO_CP,
+	KBL_DPCM_AUDIO_HS_PB,
+	KBL_DPCM_AUDIO_ECHO_REF_CP,
 	KBL_DPCM_AUDIO_REF_CP,
 	KBL_DPCM_AUDIO_DMIC_CP,
 	KBL_DPCM_AUDIO_HDMI1_PB,
@@ -485,6 +487,28 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.dpcm_capture = 1,
 		.ops = &kabylake_rt5663_fe_ops,
 	},
+	[KBL_DPCM_AUDIO_HS_PB] = {
+		.name = "Kbl Audio Headset Playback",
+		.stream_name = "Headset Audio",
+		.cpu_dai_name = "System Pin2",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Kbl Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
 	[KBL_DPCM_AUDIO_REF_CP] = {
 		.name = "Kbl Audio Reference cap",
 		.stream_name = "Wake on Voice",
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 74bbc008c965..64364de165be 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -62,6 +62,8 @@ struct kbl_codec_private {
 enum {
 	KBL_DPCM_AUDIO_PB = 0,
 	KBL_DPCM_AUDIO_CP,
+	KBL_DPCM_AUDIO_HS_PB,
+	KBL_DPCM_AUDIO_ECHO_REF_CP,
 	KBL_DPCM_AUDIO_DMIC_CP,
 	KBL_DPCM_AUDIO_HDMI1_PB,
 	KBL_DPCM_AUDIO_HDMI2_PB,
@@ -456,6 +458,28 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.dpcm_capture = 1,
 		.ops = &kabylake_rt5663_fe_ops,
 	},
+	[KBL_DPCM_AUDIO_HS_PB] = {
+		.name = "Kbl Audio Headset Playback",
+		.stream_name = "Headset Audio",
+		.cpu_dai_name = "System Pin2",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Kbl Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
 	[KBL_DPCM_AUDIO_DMIC_CP] = {
 		.name = "Kbl Audio DMIC cap",
 		.stream_name = "dmiccap",
-- 
cgit v1.2.3


From 0b06122fc8d0949a0aa7b817d98e031104802611 Mon Sep 17 00:00:00 2001
From: Naveen Manohar
Date: Wed, 2 Aug 2017 11:43:35 +0530
Subject: ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref

Modified DAPM Machine map for machine to suit the toplogy change
required to enable features:
Multi-Endpoint Playback
Echo Reference Capture

Signed-off-by: Naveen M <naveen.m@intel.com>
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c        | 6 +++---
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 6 +++---
 2 files changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 4c70d5a3028c..12b34b9a0104 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -99,12 +99,12 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 	/* CODEC BE connections */
 	{ "Left HiFi Playback", NULL, "ssp0 Tx" },
 	{ "Right HiFi Playback", NULL, "ssp0 Tx" },
-	{ "ssp0 Tx", NULL, "codec0_out" },
+	{ "ssp0 Tx", NULL, "spk_out" },
 
 	{ "AIF Playback", NULL, "ssp1 Tx" },
-	{ "ssp1 Tx", NULL, "codec1_out" },
+	{ "ssp1 Tx", NULL, "hs_out" },
 
-	{ "codec0_in", NULL, "ssp1 Rx" },
+	{ "hs_in", NULL, "ssp1 Rx" },
 	{ "ssp1 Rx", NULL, "AIF Capture" },
 
 	/* DMIC */
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 64364de165be..5ec942885b69 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -107,12 +107,12 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 	/* CODEC BE connections */
 	{ "Left HiFi Playback", NULL, "ssp0 Tx" },
 	{ "Right HiFi Playback", NULL, "ssp0 Tx" },
-	{ "ssp0 Tx", NULL, "codec0_out" },
+	{ "ssp0 Tx", NULL, "spk_out" },
 
 	{ "AIF Playback", NULL, "ssp1 Tx" },
-	{ "ssp1 Tx", NULL, "codec1_out" },
+	{ "ssp1 Tx", NULL, "hs_out" },
 
-	{ "codec0_in", NULL, "ssp1 Rx" },
+	{ "hs_in", NULL, "ssp1 Rx" },
 	{ "ssp1 Rx", NULL, "AIF Capture" },
 
 	{ "codec1_in", NULL, "ssp0 Rx" },
-- 
cgit v1.2.3


From 363d45385f403fdf60fe9f563ba87093289d7b35 Mon Sep 17 00:00:00 2001
From: Dharageswari R
Date: Wed, 2 Aug 2017 21:51:12 +0530
Subject: ASoC: Intel: Skylake: Add num of cores in dsp ops

Number of dsp cores may differ for different platforms hence
adding it in dsp ops.

Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/bxt-sst.c      | 1 -
 sound/soc/intel/skylake/skl-messages.c | 6 ++++++
 sound/soc/intel/skylake/skl-sst.c      | 2 --
 sound/soc/intel/skylake/skl.h          | 1 +
 4 files changed, 7 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index 08a2c5eb59fa..c404a42b670a 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -584,7 +584,6 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 	/* set the D0i3 check */
 	skl->ipc.ops.check_dsp_lp_on = skl_ipc_check_D0i0;
 
-	skl->cores.count = 2;
 	skl->boot_complete = false;
 	init_waitqueue_head(&skl->boot_wait);
 	INIT_DELAYED_WORK(&skl->d0i3.work, bxt_set_dsp_D0i3);
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index c5b4e0a9946c..7892e445e87b 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -201,6 +201,7 @@ static struct skl_dsp_loader_ops bxt_get_loader_ops(void)
 static const struct skl_dsp_ops dsp_ops[] = {
 	{
 		.id = 0x9d70,
+		.num_cores = 2,
 		.loader_ops = skl_get_loader_ops,
 		.init = skl_sst_dsp_init,
 		.init_fw = skl_sst_init_fw,
@@ -208,6 +209,7 @@ static const struct skl_dsp_ops dsp_ops[] = {
 	},
 	{
 		.id = 0x9d71,
+		.num_cores = 2,
 		.loader_ops = skl_get_loader_ops,
 		.init = kbl_sst_dsp_init,
 		.init_fw = skl_sst_init_fw,
@@ -215,6 +217,7 @@ static const struct skl_dsp_ops dsp_ops[] = {
 	},
 	{
 		.id = 0x5a98,
+		.num_cores = 2,
 		.loader_ops = bxt_get_loader_ops,
 		.init = bxt_sst_dsp_init,
 		.init_fw = bxt_sst_init_fw,
@@ -222,6 +225,7 @@ static const struct skl_dsp_ops dsp_ops[] = {
 	},
 	{
 		.id = 0x3198,
+		.num_cores = 2,
 		.loader_ops = bxt_get_loader_ops,
 		.init = bxt_sst_dsp_init,
 		.init_fw = bxt_sst_init_fw,
@@ -275,6 +279,8 @@ int skl_init_dsp(struct skl *skl)
 		return ret;
 
 	skl->skl_sst->dsp_ops = ops;
+	skl->skl_sst->cores.count = ops->num_cores;
+
 	dev_dbg(bus->dev, "dsp registration status=%d\n", ret);
 
 	return ret;
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index aba9ea11ac74..fd99e74cda03 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -563,8 +563,6 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 
 	sst->fw_ops = skl_fw_ops;
 
-	skl->cores.count = 2;
-
 	return 0;
 }
 EXPORT_SYMBOL_GPL(skl_sst_dsp_init);
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index a6b134b4c037..20017ee4685c 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -90,6 +90,7 @@ struct skl_machine_pdata {
 
 struct skl_dsp_ops {
 	int id;
+	unsigned int num_cores;
 	struct skl_dsp_loader_ops (*loader_ops)(void);
 	int (*init)(struct device *dev, void __iomem *mmio_base,
 			int irq, const char *fw_name,
-- 
cgit v1.2.3


From f0a550a88373b0b076785e059cca15f32ae9ffeb Mon Sep 17 00:00:00 2001
From: Guneshwor Singh
Date: Wed, 2 Aug 2017 21:51:13 +0530
Subject: ASoC: Intel: Skylake: Use num_core to allocate instead of macro

For different platforms, number of dsp cores can vary. So instead of
creating array of size SKL_DSP_CORES_MAX, use num_core in dsp_ops() of
the respective platform to allocate core usage counts and states.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 18 +++++++++++++++++-
 sound/soc/intel/skylake/skl-sst-dsp.c  |  2 +-
 sound/soc/intel/skylake/skl-sst-ipc.h  |  6 ++----
 3 files changed, 20 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 7892e445e87b..c6b60ee5b196 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -253,6 +253,7 @@ int skl_init_dsp(struct skl *skl)
 	struct skl_dsp_loader_ops loader_ops;
 	int irq = bus->irq;
 	const struct skl_dsp_ops *ops;
+	struct skl_dsp_cores *cores;
 	int ret;
 
 	/* enable ppcap interrupt */
@@ -279,7 +280,19 @@ int skl_init_dsp(struct skl *skl)
 		return ret;
 
 	skl->skl_sst->dsp_ops = ops;
-	skl->skl_sst->cores.count = ops->num_cores;
+	cores = &skl->skl_sst->cores;
+	cores->count = ops->num_cores;
+
+	cores->state = kcalloc(cores->count, sizeof(*cores->state), GFP_KERNEL);
+	if (!cores->state)
+		return -ENOMEM;
+
+	cores->usage_count = kcalloc(cores->count, sizeof(*cores->usage_count),
+				     GFP_KERNEL);
+	if (!cores->usage_count) {
+		kfree(cores->state);
+		return -ENOMEM;
+	}
 
 	dev_dbg(bus->dev, "dsp registration status=%d\n", ret);
 
@@ -297,6 +310,9 @@ int skl_free_dsp(struct skl *skl)
 
 	ctx->dsp_ops->cleanup(bus->dev, ctx);
 
+	kfree(ctx->cores.state);
+	kfree(ctx->cores.usage_count);
+
 	if (ctx->dsp->addr.lpe)
 		iounmap(ctx->dsp->addr.lpe);
 
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index 08332723c700..c2ba5ff1632c 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -47,7 +47,7 @@ void skl_dsp_init_core_state(struct sst_dsp *ctx)
 	skl->cores.state[SKL_DSP_CORE0_ID] = SKL_DSP_RUNNING;
 	skl->cores.usage_count[SKL_DSP_CORE0_ID] = 1;
 
-	for (i = SKL_DSP_CORE0_ID + 1; i < SKL_DSP_CORES_MAX; i++) {
+	for (i = SKL_DSP_CORE0_ID + 1; i < skl->cores.count; i++) {
 		skl->cores.state[i] = SKL_DSP_RESET;
 		skl->cores.usage_count[i] = 0;
 	}
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h
index e057da2713c6..e485e8f5ea2c 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.h
+++ b/sound/soc/intel/skylake/skl-sst-ipc.h
@@ -44,12 +44,10 @@ struct skl_ipc_header {
 	u32 extension;
 };
 
-#define SKL_DSP_CORES_MAX  2
-
 struct skl_dsp_cores {
 	unsigned int count;
-	enum skl_dsp_states state[SKL_DSP_CORES_MAX];
-	int usage_count[SKL_DSP_CORES_MAX];
+	enum skl_dsp_states *state;
+	int *usage_count;
 };
 
 /**
-- 
cgit v1.2.3


From 4147a6e59a5c0b9cb0eeaa3e1f39fd9da491f35a Mon Sep 17 00:00:00 2001
From: Pardha Saradhi K
Date: Wed, 2 Aug 2017 21:51:14 +0530
Subject: ASoC: Intel: Skylake: Add dsp cores management

This patch adds support to manage additional cores on a demand basis.
For instance, if module is set to run on certain core, the particular
core is powered up in module init. The same is again powered down in
module unload.

Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 26 ++++++++++++++++++++++----
 1 file changed, 22 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 8c3ce08f366d..350d2387f737 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -596,24 +596,35 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 		if (mconfig->id.pvt_id < 0)
 			return ret;
 		skl_tplg_set_module_init_data(w);
+
+		ret = skl_dsp_get_core(ctx->dsp, mconfig->core_id);
+		if (ret < 0) {
+			dev_err(ctx->dev, "Failed to wake up core %d ret=%d\n",
+						mconfig->core_id, ret);
+			return ret;
+		}
+
 		ret = skl_init_module(ctx, mconfig);
 		if (ret < 0) {
 			skl_put_pvt_id(ctx, uuid_mod, &mconfig->id.pvt_id);
-			return ret;
+			goto err;
 		}
 		skl_tplg_alloc_pipe_mcps(skl, mconfig);
 		ret = skl_tplg_set_module_params(w, ctx);
 		if (ret < 0)
-			return ret;
+			goto err;
 	}
 
 	return 0;
+err:
+	skl_dsp_put_core(ctx->dsp, mconfig->core_id);
+	return ret;
 }
 
 static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
 	 struct skl_pipe *pipe)
 {
-	int ret;
+	int ret = 0;
 	struct skl_pipe_module *w_module = NULL;
 	struct skl_module_cfg *mconfig = NULL;
 
@@ -630,10 +641,17 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
 				return -EIO;
 		}
 		skl_put_pvt_id(ctx, uuid_mod, &mconfig->id.pvt_id);
+
+		ret = skl_dsp_put_core(ctx->dsp, mconfig->core_id);
+		if (ret < 0) {
+			/* don't return; continue with other modules */
+			dev_err(ctx->dev, "Failed to sleep core %d ret=%d\n",
+				mconfig->core_id, ret);
+		}
 	}
 
 	/* no modules to unload in this path, so return */
-	return 0;
+	return ret;
 }
 
 /*
-- 
cgit v1.2.3


From a838dcc286754f6ea6bcbee8eb76a5d626642fa7 Mon Sep 17 00:00:00 2001
From: Guneshwor Singh
Date: Wed, 2 Aug 2017 21:51:15 +0530
Subject: ASoC: Intel: cnl: Add cnl dsp functions and registers

This adds Cannonlake specific registers and support for CNL dsp related
library functions for programming the registers to power up/down dsp cores,
set/unset reset states for each core, enable/disable ipc interrupts and few
wrappers to be called from elsewhere.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/cnl-sst-dsp.c | 274 ++++++++++++++++++++++++++++++++++
 sound/soc/intel/skylake/cnl-sst-dsp.h | 106 +++++++++++++
 2 files changed, 380 insertions(+)
 create mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c
 create mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/cnl-sst-dsp.c b/sound/soc/intel/skylake/cnl-sst-dsp.c
new file mode 100644
index 000000000000..2f8326707c21
--- /dev/null
+++ b/sound/soc/intel/skylake/cnl-sst-dsp.c
@@ -0,0 +1,274 @@
+/*
+ * cnl-sst-dsp.c - CNL SST library generic function
+ *
+ * Copyright (C) 2016-17, Intel Corporation.
+ * Author: Guneshwor Singh <guneshwor.o.singh@intel.com>
+ *
+ * Modified from:
+ *	SKL SST library generic function
+ *	Copyright (C) 2014-15, Intel Corporation.
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+#include <linux/device.h>
+#include "../common/sst-dsp.h"
+#include "../common/sst-ipc.h"
+#include "../common/sst-dsp-priv.h"
+#include "cnl-sst-dsp.h"
+
+/* various timeout values */
+#define CNL_DSP_PU_TO		50
+#define CNL_DSP_PD_TO		50
+#define CNL_DSP_RESET_TO	50
+
+static int
+cnl_dsp_core_set_reset_state(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	/* update bits */
+	sst_dsp_shim_update_bits_unlocked(ctx,
+			CNL_ADSP_REG_ADSPCS, CNL_ADSPCS_CRST(core_mask),
+			CNL_ADSPCS_CRST(core_mask));
+
+	/* poll with timeout to check if operation successful */
+	return sst_dsp_register_poll(ctx,
+			CNL_ADSP_REG_ADSPCS,
+			CNL_ADSPCS_CRST(core_mask),
+			CNL_ADSPCS_CRST(core_mask),
+			CNL_DSP_RESET_TO,
+			"Set reset");
+}
+
+static int
+cnl_dsp_core_unset_reset_state(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	/* update bits */
+	sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS,
+					CNL_ADSPCS_CRST(core_mask), 0);
+
+	/* poll with timeout to check if operation successful */
+	return sst_dsp_register_poll(ctx,
+			CNL_ADSP_REG_ADSPCS,
+			CNL_ADSPCS_CRST(core_mask),
+			0,
+			CNL_DSP_RESET_TO,
+			"Unset reset");
+}
+
+static bool is_cnl_dsp_core_enable(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	int val;
+	bool is_enable;
+
+	val = sst_dsp_shim_read_unlocked(ctx, CNL_ADSP_REG_ADSPCS);
+
+	is_enable = (val & CNL_ADSPCS_CPA(core_mask)) &&
+			(val & CNL_ADSPCS_SPA(core_mask)) &&
+			!(val & CNL_ADSPCS_CRST(core_mask)) &&
+			!(val & CNL_ADSPCS_CSTALL(core_mask));
+
+	dev_dbg(ctx->dev, "DSP core(s) enabled? %d: core_mask %#x\n",
+		is_enable, core_mask);
+
+	return is_enable;
+}
+
+static int cnl_dsp_reset_core(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	/* stall core */
+	sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS,
+			CNL_ADSPCS_CSTALL(core_mask),
+			CNL_ADSPCS_CSTALL(core_mask));
+
+	/* set reset state */
+	return cnl_dsp_core_set_reset_state(ctx, core_mask);
+}
+
+static int cnl_dsp_start_core(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	int ret;
+
+	/* unset reset state */
+	ret = cnl_dsp_core_unset_reset_state(ctx, core_mask);
+	if (ret < 0)
+		return ret;
+
+	/* run core */
+	sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS,
+				CNL_ADSPCS_CSTALL(core_mask), 0);
+
+	if (!is_cnl_dsp_core_enable(ctx, core_mask)) {
+		cnl_dsp_reset_core(ctx, core_mask);
+		dev_err(ctx->dev, "DSP core mask %#x enable failed\n",
+			core_mask);
+		ret = -EIO;
+	}
+
+	return ret;
+}
+
+static int cnl_dsp_core_power_up(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	/* update bits */
+	sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS,
+					  CNL_ADSPCS_SPA(core_mask),
+					  CNL_ADSPCS_SPA(core_mask));
+
+	/* poll with timeout to check if operation successful */
+	return sst_dsp_register_poll(ctx, CNL_ADSP_REG_ADSPCS,
+				    CNL_ADSPCS_CPA(core_mask),
+				    CNL_ADSPCS_CPA(core_mask),
+				    CNL_DSP_PU_TO,
+				    "Power up");
+}
+
+static int cnl_dsp_core_power_down(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	/* update bits */
+	sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS,
+					CNL_ADSPCS_SPA(core_mask), 0);
+
+	/* poll with timeout to check if operation successful */
+	return sst_dsp_register_poll(ctx,
+			CNL_ADSP_REG_ADSPCS,
+			CNL_ADSPCS_CPA(core_mask),
+			0,
+			CNL_DSP_PD_TO,
+			"Power down");
+}
+
+int cnl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	int ret;
+
+	/* power up */
+	ret = cnl_dsp_core_power_up(ctx, core_mask);
+	if (ret < 0) {
+		dev_dbg(ctx->dev, "DSP core mask %#x power up failed",
+			core_mask);
+		return ret;
+	}
+
+	return cnl_dsp_start_core(ctx, core_mask);
+}
+
+int cnl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core_mask)
+{
+	int ret;
+
+	ret = cnl_dsp_reset_core(ctx, core_mask);
+	if (ret < 0) {
+		dev_err(ctx->dev, "DSP core mask %#x reset failed\n",
+			core_mask);
+		return ret;
+	}
+
+	/* power down core*/
+	ret = cnl_dsp_core_power_down(ctx, core_mask);
+	if (ret < 0) {
+		dev_err(ctx->dev, "DSP core mask %#x power down failed\n",
+			core_mask);
+		return ret;
+	}
+
+	if (is_cnl_dsp_core_enable(ctx, core_mask)) {
+		dev_err(ctx->dev, "DSP core mask %#x disable failed\n",
+			core_mask);
+		ret = -EIO;
+	}
+
+	return ret;
+}
+
+irqreturn_t cnl_dsp_sst_interrupt(int irq, void *dev_id)
+{
+	struct sst_dsp *ctx = dev_id;
+	u32 val;
+	irqreturn_t ret = IRQ_NONE;
+
+	spin_lock(&ctx->spinlock);
+
+	val = sst_dsp_shim_read_unlocked(ctx, CNL_ADSP_REG_ADSPIS);
+	ctx->intr_status = val;
+
+	if (val == 0xffffffff) {
+		spin_unlock(&ctx->spinlock);
+		return IRQ_NONE;
+	}
+
+	if (val & CNL_ADSPIS_IPC) {
+		cnl_ipc_int_disable(ctx);
+		ret = IRQ_WAKE_THREAD;
+	}
+
+	spin_unlock(&ctx->spinlock);
+
+	return ret;
+}
+
+void cnl_dsp_free(struct sst_dsp *dsp)
+{
+	cnl_ipc_int_disable(dsp);
+
+	free_irq(dsp->irq, dsp);
+	cnl_ipc_op_int_disable(dsp);
+	cnl_dsp_disable_core(dsp, SKL_DSP_CORE0_MASK);
+}
+EXPORT_SYMBOL_GPL(cnl_dsp_free);
+
+void cnl_ipc_int_enable(struct sst_dsp *ctx)
+{
+	sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_ADSPIC,
+				 CNL_ADSPIC_IPC, CNL_ADSPIC_IPC);
+}
+
+void cnl_ipc_int_disable(struct sst_dsp *ctx)
+{
+	sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPIC,
+					  CNL_ADSPIC_IPC, 0);
+}
+
+void cnl_ipc_op_int_enable(struct sst_dsp *ctx)
+{
+	/* enable IPC DONE interrupt */
+	sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL,
+				 CNL_ADSP_REG_HIPCCTL_DONE,
+				 CNL_ADSP_REG_HIPCCTL_DONE);
+
+	/* enable IPC BUSY interrupt */
+	sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL,
+				 CNL_ADSP_REG_HIPCCTL_BUSY,
+				 CNL_ADSP_REG_HIPCCTL_BUSY);
+}
+
+void cnl_ipc_op_int_disable(struct sst_dsp *ctx)
+{
+	/* disable IPC DONE interrupt */
+	sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL,
+				 CNL_ADSP_REG_HIPCCTL_DONE, 0);
+
+	/* disable IPC BUSY interrupt */
+	sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL,
+				 CNL_ADSP_REG_HIPCCTL_BUSY, 0);
+}
+
+bool cnl_ipc_int_status(struct sst_dsp *ctx)
+{
+	return sst_dsp_shim_read_unlocked(ctx, CNL_ADSP_REG_ADSPIS) &
+							CNL_ADSPIS_IPC;
+}
+
+void cnl_ipc_free(struct sst_generic_ipc *ipc)
+{
+	cnl_ipc_op_int_disable(ipc->dsp);
+	sst_ipc_fini(ipc);
+}
diff --git a/sound/soc/intel/skylake/cnl-sst-dsp.h b/sound/soc/intel/skylake/cnl-sst-dsp.h
new file mode 100644
index 000000000000..2fe41b137a2c
--- /dev/null
+++ b/sound/soc/intel/skylake/cnl-sst-dsp.h
@@ -0,0 +1,106 @@
+/*
+ * Cannonlake SST DSP Support
+ *
+ * Copyright (C) 2016-17, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef __CNL_SST_DSP_H__
+#define __CNL_SST_DSP_H__
+
+struct sst_dsp;
+struct skl_sst;
+struct sst_dsp_device;
+struct sst_generic_ipc;
+
+/* Intel HD Audio General DSP Registers */
+#define CNL_ADSP_GEN_BASE		0x0
+#define CNL_ADSP_REG_ADSPCS		(CNL_ADSP_GEN_BASE + 0x04)
+#define CNL_ADSP_REG_ADSPIC		(CNL_ADSP_GEN_BASE + 0x08)
+#define CNL_ADSP_REG_ADSPIS		(CNL_ADSP_GEN_BASE + 0x0c)
+
+/* Intel HD Audio Inter-Processor Communication Registers */
+#define CNL_ADSP_IPC_BASE               0xc0
+#define CNL_ADSP_REG_HIPCTDR            (CNL_ADSP_IPC_BASE + 0x00)
+#define CNL_ADSP_REG_HIPCTDA            (CNL_ADSP_IPC_BASE + 0x04)
+#define CNL_ADSP_REG_HIPCTDD            (CNL_ADSP_IPC_BASE + 0x08)
+#define CNL_ADSP_REG_HIPCIDR            (CNL_ADSP_IPC_BASE + 0x10)
+#define CNL_ADSP_REG_HIPCIDA            (CNL_ADSP_IPC_BASE + 0x14)
+#define CNL_ADSP_REG_HIPCIDD            (CNL_ADSP_IPC_BASE + 0x18)
+#define CNL_ADSP_REG_HIPCCTL            (CNL_ADSP_IPC_BASE + 0x28)
+
+/* HIPCTDR */
+#define CNL_ADSP_REG_HIPCTDR_BUSY	BIT(31)
+
+/* HIPCTDA */
+#define CNL_ADSP_REG_HIPCTDA_DONE	BIT(31)
+
+/* HIPCIDR */
+#define CNL_ADSP_REG_HIPCIDR_BUSY	BIT(31)
+
+/* HIPCIDA */
+#define CNL_ADSP_REG_HIPCIDA_DONE	BIT(31)
+
+/* CNL HIPCCTL */
+#define CNL_ADSP_REG_HIPCCTL_DONE	BIT(1)
+#define CNL_ADSP_REG_HIPCCTL_BUSY	BIT(0)
+
+/* CNL HIPCT */
+#define CNL_ADSP_REG_HIPCT_BUSY		BIT(31)
+
+/* Intel HD Audio SRAM Window 1 */
+#define CNL_ADSP_SRAM1_BASE		0xa0000
+
+#define CNL_ADSP_MMIO_LEN		0x10000
+
+#define CNL_ADSP_W0_STAT_SZ		0x1000
+
+#define CNL_ADSP_W0_UP_SZ		0x1000
+
+#define CNL_ADSP_W1_SZ			0x1000
+
+#define CNL_FW_STS_MASK			0xf
+
+#define CNL_ADSPIC_IPC			0x1
+#define CNL_ADSPIS_IPC			0x1
+
+#define CNL_DSP_CORES		4
+#define CNL_DSP_CORES_MASK	((1 << CNL_DSP_CORES) - 1)
+
+/* core reset - asserted high */
+#define CNL_ADSPCS_CRST_SHIFT	0
+#define CNL_ADSPCS_CRST(x)	(x << CNL_ADSPCS_CRST_SHIFT)
+
+/* core run/stall - when set to 1 core is stalled */
+#define CNL_ADSPCS_CSTALL_SHIFT	8
+#define CNL_ADSPCS_CSTALL(x)	(x << CNL_ADSPCS_CSTALL_SHIFT)
+
+/* set power active - when set to 1 turn core on */
+#define CNL_ADSPCS_SPA_SHIFT	16
+#define CNL_ADSPCS_SPA(x)	(x << CNL_ADSPCS_SPA_SHIFT)
+
+/* current power active - power status of cores, set by hardware */
+#define CNL_ADSPCS_CPA_SHIFT	24
+#define CNL_ADSPCS_CPA(x)	(x << CNL_ADSPCS_CPA_SHIFT)
+
+int cnl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core);
+int cnl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core);
+irqreturn_t cnl_dsp_sst_interrupt(int irq, void *dev_id);
+void cnl_dsp_free(struct sst_dsp *dsp);
+
+void cnl_ipc_int_enable(struct sst_dsp *ctx);
+void cnl_ipc_int_disable(struct sst_dsp *ctx);
+void cnl_ipc_op_int_enable(struct sst_dsp *ctx);
+void cnl_ipc_op_int_disable(struct sst_dsp *ctx);
+bool cnl_ipc_int_status(struct sst_dsp *ctx);
+void cnl_ipc_free(struct sst_generic_ipc *ipc);
+
+#endif /*__CNL_SST_DSP_H__*/
-- 
cgit v1.2.3


From 2eed1b024a11b497766604ee8f95e22711859af2 Mon Sep 17 00:00:00 2001
From: Guneshwor Singh
Date: Wed, 2 Aug 2017 21:51:16 +0530
Subject: ASoC: Intel: Skylake: Move platform specific init to platform
 dsp_init()

Move ipc_init() from helper function to respective platform's dsp_init()
as ipc_init() per platform can be different.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/bxt-sst.c       | 4 ++++
 sound/soc/intel/skylake/skl-sst-utils.c | 6 +-----
 sound/soc/intel/skylake/skl-sst.c       | 4 ++++
 3 files changed, 9 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index c404a42b670a..71042aceb25f 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -581,6 +581,10 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 	sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ),
 			SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
 
+	ret = skl_ipc_init(dev, skl);
+	if (ret)
+		return ret;
+
 	/* set the D0i3 check */
 	skl->ipc.ops.check_dsp_lp_on = skl_ipc_check_D0i0;
 
diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c
index 81ee251881b4..369ef7ce981c 100644
--- a/sound/soc/intel/skylake/skl-sst-utils.c
+++ b/sound/soc/intel/skylake/skl-sst-utils.c
@@ -368,7 +368,6 @@ int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name,
 {
 	struct skl_sst *skl;
 	struct sst_dsp *sst;
-	int ret;
 
 	skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL);
 	if (skl == NULL)
@@ -388,15 +387,12 @@ int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name,
 	sst->dsp_ops = dsp_ops;
 	init_waitqueue_head(&skl->mod_load_wait);
 	INIT_LIST_HEAD(&sst->module_list);
-	ret = skl_ipc_init(dev, skl);
-	if (ret)
-		return ret;
 
 	skl->is_first_boot = true;
 	if (dsp)
 		*dsp = skl;
 
-	return ret;
+	return 0;
 }
 
 int skl_prepare_lib_load(struct skl_sst *skl, struct skl_lib_info *linfo,
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index fd99e74cda03..6cc586d3f03b 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -561,6 +561,10 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 	sst_dsp_mailbox_init(sst, (SKL_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ),
 			SKL_ADSP_W0_UP_SZ, SKL_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
 
+	ret = skl_ipc_init(dev, skl);
+	if (ret)
+		return ret;
+
 	sst->fw_ops = skl_fw_ops;
 
 	return 0;
-- 
cgit v1.2.3


From 4bdb04f0e1b7600b87a92b4f98ba5490ceffdf4f Mon Sep 17 00:00:00 2001
From: Guneshwor Singh
Date: Wed, 2 Aug 2017 21:51:17 +0530
Subject: ASoC: Intel: cnl: Unstatify common ipc functions

Common ipc functions can be reused for cnl, so make them non-static.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-sst-ipc.c | 6 +++---
 sound/soc/intel/skylake/skl-sst-ipc.h | 6 ++++++
 2 files changed, 9 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index 498b15345b1a..5234fafb758a 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -283,7 +283,7 @@ enum skl_ipc_module_msg {
 	IPC_MOD_SET_D0IX = 8
 };
 
-static void skl_ipc_tx_data_copy(struct ipc_message *msg, char *tx_data,
+void skl_ipc_tx_data_copy(struct ipc_message *msg, char *tx_data,
 		size_t tx_size)
 {
 	if (tx_size)
@@ -347,7 +347,7 @@ out:
 
 }
 
-static int skl_ipc_process_notification(struct sst_generic_ipc *ipc,
+int skl_ipc_process_notification(struct sst_generic_ipc *ipc,
 		struct skl_ipc_header header)
 {
 	struct skl_sst *skl = container_of(ipc, struct skl_sst, ipc);
@@ -406,7 +406,7 @@ static int skl_ipc_set_reply_error_code(u32 reply)
 	}
 }
 
-static void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
+void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
 		struct skl_ipc_header header)
 {
 	struct ipc_message *msg;
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h
index e485e8f5ea2c..55f2d2ce09df 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.h
+++ b/sound/soc/intel/skylake/skl-sst-ipc.h
@@ -212,4 +212,10 @@ void skl_ipc_free(struct sst_generic_ipc *ipc);
 int skl_ipc_init(struct device *dev, struct skl_sst *skl);
 void skl_clear_module_cnt(struct sst_dsp *ctx);
 
+void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
+		struct skl_ipc_header header);
+int skl_ipc_process_notification(struct sst_generic_ipc *ipc,
+		struct skl_ipc_header header);
+void skl_ipc_tx_data_copy(struct ipc_message *msg, char *tx_data,
+		size_t tx_size);
 #endif /* __SKL_IPC_H */
-- 
cgit v1.2.3


From cb6a552846297094aac942a588fe7ed7922a6c11 Mon Sep 17 00:00:00 2001
From: Guneshwor Singh
Date: Wed, 2 Aug 2017 21:51:18 +0530
Subject: ASoC: Intel: cnl: Add sst library functions for cnl platform

This adds the necessary DSP functions specific for the Cannonlake platform
which includes firmware download using host DMA, DO/D3 handlers, irq_thread
handlers and sst ops.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/Makefile      |   5 +-
 sound/soc/intel/skylake/cnl-sst-dsp.h |   6 +
 sound/soc/intel/skylake/cnl-sst.c     | 495 ++++++++++++++++++++++++++++++++++
 3 files changed, 504 insertions(+), 2 deletions(-)
 create mode 100644 sound/soc/intel/skylake/cnl-sst.c

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index e7d77722d560..3380deb81015 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -8,7 +8,8 @@ endif
 obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
 
 # Skylake IPC Support
-snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \
-		skl-sst.o bxt-sst.o skl-sst-utils.o
+snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o cnl-sst-dsp.o \
+		skl-sst-cldma.o skl-sst.o bxt-sst.o cnl-sst.o \
+		skl-sst-utils.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o
diff --git a/sound/soc/intel/skylake/cnl-sst-dsp.h b/sound/soc/intel/skylake/cnl-sst-dsp.h
index 2fe41b137a2c..09bd218df5c4 100644
--- a/sound/soc/intel/skylake/cnl-sst-dsp.h
+++ b/sound/soc/intel/skylake/cnl-sst-dsp.h
@@ -103,4 +103,10 @@ void cnl_ipc_op_int_disable(struct sst_dsp *ctx);
 bool cnl_ipc_int_status(struct sst_dsp *ctx);
 void cnl_ipc_free(struct sst_generic_ipc *ipc);
 
+int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+		     const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+		     struct skl_sst **dsp);
+int cnl_sst_init_fw(struct device *dev, struct skl_sst *ctx);
+void cnl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
+
 #endif /*__CNL_SST_DSP_H__*/
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
new file mode 100644
index 000000000000..7dcfe66a20e2
--- /dev/null
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -0,0 +1,495 @@
+/*
+ * cnl-sst.c - DSP library functions for CNL platform
+ *
+ * Copyright (C) 2016-17, Intel Corporation.
+ *
+ * Author: Guneshwor Singh <guneshwor.o.singh@intel.com>
+ *
+ * Modified from:
+ *	HDA DSP library functions for SKL platform
+ *	Copyright (C) 2014-15, Intel Corporation.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/firmware.h>
+#include <linux/device.h>
+
+#include "../common/sst-dsp.h"
+#include "../common/sst-dsp-priv.h"
+#include "../common/sst-ipc.h"
+#include "cnl-sst-dsp.h"
+#include "skl-sst-dsp.h"
+#include "skl-sst-ipc.h"
+
+#define CNL_FW_ROM_INIT		0x1
+#define CNL_FW_INIT		0x5
+#define CNL_IPC_PURGE		0x01004000
+#define CNL_INIT_TIMEOUT	300
+#define CNL_BASEFW_TIMEOUT	3000
+
+#define CNL_ADSP_SRAM0_BASE	0x80000
+
+/* Firmware status window */
+#define CNL_ADSP_FW_STATUS	CNL_ADSP_SRAM0_BASE
+#define CNL_ADSP_ERROR_CODE	(CNL_ADSP_FW_STATUS + 0x4)
+
+#define CNL_INSTANCE_ID		0
+#define CNL_BASE_FW_MODULE_ID	0
+#define CNL_ADSP_FW_HDR_OFFSET	0x2000
+#define CNL_ROM_CTRL_DMA_ID	0x9
+
+static int cnl_prepare_fw(struct sst_dsp *ctx, const void *fwdata, u32 fwsize)
+{
+
+	int ret, stream_tag;
+
+	stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab);
+	if (stream_tag <= 0) {
+		dev_err(ctx->dev, "dma prepare failed: 0%#x\n", stream_tag);
+		return stream_tag;
+	}
+
+	ctx->dsp_ops.stream_tag = stream_tag;
+	memcpy(ctx->dmab.area, fwdata, fwsize);
+
+	/* purge FW request */
+	sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR,
+			   CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE |
+			   ((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID)));
+
+	ret = cnl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK);
+	if (ret < 0) {
+		dev_err(ctx->dev, "dsp boot core failed ret: %d\n", ret);
+		ret = -EIO;
+		goto base_fw_load_failed;
+	}
+
+	/* enable interrupt */
+	cnl_ipc_int_enable(ctx);
+	cnl_ipc_op_int_enable(ctx);
+
+	ret = sst_dsp_register_poll(ctx, CNL_ADSP_FW_STATUS, CNL_FW_STS_MASK,
+				    CNL_FW_ROM_INIT, CNL_INIT_TIMEOUT,
+				    "rom load");
+	if (ret < 0) {
+		dev_err(ctx->dev, "rom init timeout, ret: %d\n", ret);
+		goto base_fw_load_failed;
+	}
+
+	return 0;
+
+base_fw_load_failed:
+	ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag);
+	cnl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
+
+	return ret;
+}
+
+static int sst_transfer_fw_host_dma(struct sst_dsp *ctx)
+{
+	int ret;
+
+	ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag);
+	ret = sst_dsp_register_poll(ctx, CNL_ADSP_FW_STATUS, CNL_FW_STS_MASK,
+				    CNL_FW_INIT, CNL_BASEFW_TIMEOUT,
+				    "firmware boot");
+
+	ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag);
+	ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag);
+
+	return ret;
+}
+
+static int cnl_load_base_firmware(struct sst_dsp *ctx)
+{
+	struct firmware stripped_fw;
+	struct skl_sst *cnl = ctx->thread_context;
+	int ret;
+
+	if (!ctx->fw) {
+		ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev);
+		if (ret < 0) {
+			dev_err(ctx->dev, "request firmware failed: %d\n", ret);
+			goto cnl_load_base_firmware_failed;
+		}
+	}
+
+	/* parse uuids if first boot */
+	if (cnl->is_first_boot) {
+		ret = snd_skl_parse_uuids(ctx, ctx->fw,
+					  CNL_ADSP_FW_HDR_OFFSET, 0);
+		if (ret < 0)
+			goto cnl_load_base_firmware_failed;
+	}
+
+	stripped_fw.data = ctx->fw->data;
+	stripped_fw.size = ctx->fw->size;
+	skl_dsp_strip_extended_manifest(&stripped_fw);
+
+	ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
+	if (ret < 0) {
+		dev_err(ctx->dev, "prepare firmware failed: %d\n", ret);
+		goto cnl_load_base_firmware_failed;
+	}
+
+	ret = sst_transfer_fw_host_dma(ctx);
+	if (ret < 0) {
+		dev_err(ctx->dev, "transfer firmware failed: %d\n", ret);
+		cnl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
+		goto cnl_load_base_firmware_failed;
+	}
+
+	ret = wait_event_timeout(cnl->boot_wait, cnl->boot_complete,
+				 msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+	if (ret == 0) {
+		dev_err(ctx->dev, "FW ready timed-out\n");
+		cnl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK);
+		ret = -EIO;
+		goto cnl_load_base_firmware_failed;
+	}
+
+	cnl->fw_loaded = true;
+
+	return 0;
+
+cnl_load_base_firmware_failed:
+	release_firmware(ctx->fw);
+	ctx->fw = NULL;
+
+	return ret;
+}
+
+static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id)
+{
+	struct skl_sst *cnl = ctx->thread_context;
+	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
+	struct skl_ipc_dxstate_info dx;
+	int ret;
+
+	if (!cnl->fw_loaded) {
+		cnl->boot_complete = false;
+		ret = cnl_load_base_firmware(ctx);
+		if (ret < 0) {
+			dev_err(ctx->dev, "fw reload failed: %d\n", ret);
+			return ret;
+		}
+
+		cnl->cores.state[core_id] = SKL_DSP_RUNNING;
+		return ret;
+	}
+
+	ret = cnl_dsp_enable_core(ctx, core_mask);
+	if (ret < 0) {
+		dev_err(ctx->dev, "enable dsp core %d failed: %d\n",
+			core_id, ret);
+		goto err;
+	}
+
+	if (core_id == SKL_DSP_CORE0_ID) {
+		/* enable interrupt */
+		cnl_ipc_int_enable(ctx);
+		cnl_ipc_op_int_enable(ctx);
+		cnl->boot_complete = false;
+
+		ret = wait_event_timeout(cnl->boot_wait, cnl->boot_complete,
+					 msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+		if (ret == 0) {
+			dev_err(ctx->dev,
+				"dsp boot timeout, status=%#x error=%#x\n",
+				sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS),
+				sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE));
+			goto err;
+		}
+	} else {
+		dx.core_mask = core_mask;
+		dx.dx_mask = core_mask;
+
+		ret = skl_ipc_set_dx(&cnl->ipc, CNL_INSTANCE_ID,
+				     CNL_BASE_FW_MODULE_ID, &dx);
+		if (ret < 0) {
+			dev_err(ctx->dev, "set_dx failed, core: %d ret: %d\n",
+				core_id, ret);
+			goto err;
+		}
+	}
+	cnl->cores.state[core_id] = SKL_DSP_RUNNING;
+
+	return 0;
+err:
+	cnl_dsp_disable_core(ctx, core_mask);
+
+	return ret;
+}
+
+static int cnl_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id)
+{
+	struct skl_sst *cnl = ctx->thread_context;
+	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
+	struct skl_ipc_dxstate_info dx;
+	int ret;
+
+	dx.core_mask = core_mask;
+	dx.dx_mask = SKL_IPC_D3_MASK;
+
+	ret = skl_ipc_set_dx(&cnl->ipc, CNL_INSTANCE_ID,
+			     CNL_BASE_FW_MODULE_ID, &dx);
+	if (ret < 0) {
+		dev_err(ctx->dev,
+			"dsp core %d to d3 failed; continue reset\n",
+			core_id);
+		cnl->fw_loaded = false;
+	}
+
+	/* disable interrupts if core 0 */
+	if (core_id == SKL_DSP_CORE0_ID) {
+		skl_ipc_op_int_disable(ctx);
+		skl_ipc_int_disable(ctx);
+	}
+
+	ret = cnl_dsp_disable_core(ctx, core_mask);
+	if (ret < 0) {
+		dev_err(ctx->dev, "disable dsp core %d failed: %d\n",
+			core_id, ret);
+		return ret;
+	}
+
+	cnl->cores.state[core_id] = SKL_DSP_RESET;
+
+	return ret;
+}
+
+static unsigned int cnl_get_errno(struct sst_dsp *ctx)
+{
+	return sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE);
+}
+
+static struct skl_dsp_fw_ops cnl_fw_ops = {
+	.set_state_D0 = cnl_set_dsp_D0,
+	.set_state_D3 = cnl_set_dsp_D3,
+	.load_fw = cnl_load_base_firmware,
+	.get_fw_errcode = cnl_get_errno,
+};
+
+static struct sst_ops cnl_ops = {
+	.irq_handler = cnl_dsp_sst_interrupt,
+	.write = sst_shim32_write,
+	.read = sst_shim32_read,
+	.ram_read = sst_memcpy_fromio_32,
+	.ram_write = sst_memcpy_toio_32,
+	.free = cnl_dsp_free,
+};
+
+#define CNL_IPC_GLB_NOTIFY_RSP_SHIFT	29
+#define CNL_IPC_GLB_NOTIFY_RSP_MASK	0x1
+#define CNL_IPC_GLB_NOTIFY_RSP_TYPE(x)	(((x) >> CNL_IPC_GLB_NOTIFY_RSP_SHIFT) \
+					& CNL_IPC_GLB_NOTIFY_RSP_MASK)
+
+static irqreturn_t cnl_dsp_irq_thread_handler(int irq, void *context)
+{
+	struct sst_dsp *dsp = context;
+	struct skl_sst *cnl = sst_dsp_get_thread_context(dsp);
+	struct sst_generic_ipc *ipc = &cnl->ipc;
+	struct skl_ipc_header header = {0};
+	u32 hipcida, hipctdr, hipctdd;
+	int ipc_irq = 0;
+
+	/* here we handle ipc interrupts only */
+	if (!(dsp->intr_status & CNL_ADSPIS_IPC))
+		return IRQ_NONE;
+
+	hipcida = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCIDA);
+	hipctdr = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCTDR);
+
+	/* reply message from dsp */
+	if (hipcida & CNL_ADSP_REG_HIPCIDA_DONE) {
+		sst_dsp_shim_update_bits(dsp, CNL_ADSP_REG_HIPCCTL,
+			CNL_ADSP_REG_HIPCCTL_DONE, 0);
+
+		/* clear done bit - tell dsp operation is complete */
+		sst_dsp_shim_update_bits_forced(dsp, CNL_ADSP_REG_HIPCIDA,
+			CNL_ADSP_REG_HIPCIDA_DONE, CNL_ADSP_REG_HIPCIDA_DONE);
+
+		ipc_irq = 1;
+
+		/* unmask done interrupt */
+		sst_dsp_shim_update_bits(dsp, CNL_ADSP_REG_HIPCCTL,
+			CNL_ADSP_REG_HIPCCTL_DONE, CNL_ADSP_REG_HIPCCTL_DONE);
+	}
+
+	/* new message from dsp */
+	if (hipctdr & CNL_ADSP_REG_HIPCTDR_BUSY) {
+		hipctdd = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCTDD);
+		header.primary = hipctdr;
+		header.extension = hipctdd;
+		dev_dbg(dsp->dev, "IPC irq: Firmware respond primary:%x",
+						header.primary);
+		dev_dbg(dsp->dev, "IPC irq: Firmware respond extension:%x",
+						header.extension);
+
+		if (CNL_IPC_GLB_NOTIFY_RSP_TYPE(header.primary)) {
+			/* Handle Immediate reply from DSP Core */
+			skl_ipc_process_reply(ipc, header);
+		} else {
+			dev_dbg(dsp->dev, "IPC irq: Notification from firmware\n");
+			skl_ipc_process_notification(ipc, header);
+		}
+		/* clear busy interrupt */
+		sst_dsp_shim_update_bits_forced(dsp, CNL_ADSP_REG_HIPCTDR,
+			CNL_ADSP_REG_HIPCTDR_BUSY, CNL_ADSP_REG_HIPCTDR_BUSY);
+
+		/* set done bit to ack dsp */
+		sst_dsp_shim_update_bits_forced(dsp, CNL_ADSP_REG_HIPCTDA,
+			CNL_ADSP_REG_HIPCTDA_DONE, CNL_ADSP_REG_HIPCTDA_DONE);
+		ipc_irq = 1;
+	}
+
+	if (ipc_irq == 0)
+		return IRQ_NONE;
+
+	cnl_ipc_int_enable(dsp);
+
+	/* continue to send any remaining messages */
+	schedule_work(&ipc->kwork);
+
+	return IRQ_HANDLED;
+}
+
+static struct sst_dsp_device cnl_dev = {
+	.thread = cnl_dsp_irq_thread_handler,
+	.ops = &cnl_ops,
+};
+
+static void cnl_ipc_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg)
+{
+	struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->header);
+
+	if (msg->tx_size)
+		sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size);
+	sst_dsp_shim_write_unlocked(ipc->dsp, CNL_ADSP_REG_HIPCIDD,
+				    header->extension);
+	sst_dsp_shim_write_unlocked(ipc->dsp, CNL_ADSP_REG_HIPCIDR,
+				header->primary | CNL_ADSP_REG_HIPCIDR_BUSY);
+}
+
+static bool cnl_ipc_is_dsp_busy(struct sst_dsp *dsp)
+{
+	u32 hipcidr;
+
+	hipcidr = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCIDR);
+
+	return (hipcidr & CNL_ADSP_REG_HIPCIDR_BUSY);
+}
+
+static int cnl_ipc_init(struct device *dev, struct skl_sst *cnl)
+{
+	struct sst_generic_ipc *ipc;
+	int err;
+
+	ipc = &cnl->ipc;
+	ipc->dsp = cnl->dsp;
+	ipc->dev = dev;
+
+	ipc->tx_data_max_size = CNL_ADSP_W1_SZ;
+	ipc->rx_data_max_size = CNL_ADSP_W0_UP_SZ;
+
+	err = sst_ipc_init(ipc);
+	if (err)
+		return err;
+
+	/*
+	 * overriding tx_msg and is_dsp_busy since
+	 * ipc registers are different for cnl
+	 */
+	ipc->ops.tx_msg = cnl_ipc_tx_msg;
+	ipc->ops.tx_data_copy = skl_ipc_tx_data_copy;
+	ipc->ops.is_dsp_busy = cnl_ipc_is_dsp_busy;
+
+	return 0;
+}
+
+int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+		     const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+		     struct skl_sst **dsp)
+{
+	struct skl_sst *cnl;
+	struct sst_dsp *sst;
+	int ret;
+
+	ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &cnl_dev);
+	if (ret < 0) {
+		dev_err(dev, "%s: no device\n", __func__);
+		return ret;
+	}
+
+	cnl = *dsp;
+	sst = cnl->dsp;
+	sst->fw_ops = cnl_fw_ops;
+	sst->addr.lpe = mmio_base;
+	sst->addr.shim = mmio_base;
+	sst->addr.sram0_base = CNL_ADSP_SRAM0_BASE;
+	sst->addr.sram1_base = CNL_ADSP_SRAM1_BASE;
+	sst->addr.w0_stat_sz = CNL_ADSP_W0_STAT_SZ;
+	sst->addr.w0_up_sz = CNL_ADSP_W0_UP_SZ;
+
+	sst_dsp_mailbox_init(sst, (CNL_ADSP_SRAM0_BASE + CNL_ADSP_W0_STAT_SZ),
+			     CNL_ADSP_W0_UP_SZ, CNL_ADSP_SRAM1_BASE,
+			     CNL_ADSP_W1_SZ);
+
+	ret = cnl_ipc_init(dev, cnl);
+	if (ret)
+		return ret;
+
+	cnl->boot_complete = false;
+	init_waitqueue_head(&cnl->boot_wait);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(cnl_sst_dsp_init);
+
+int cnl_sst_init_fw(struct device *dev, struct skl_sst *ctx)
+{
+	int ret;
+	struct sst_dsp *sst = ctx->dsp;
+
+	ret = ctx->dsp->fw_ops.load_fw(sst);
+	if (ret < 0) {
+		dev_err(dev, "load base fw failed: %d", ret);
+		return ret;
+	}
+
+	skl_dsp_init_core_state(sst);
+
+	ctx->is_first_boot = false;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(cnl_sst_init_fw);
+
+void cnl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
+{
+	if (ctx->dsp->fw)
+		release_firmware(ctx->dsp->fw);
+
+	skl_freeup_uuid_list(ctx);
+	cnl_ipc_free(&ctx->ipc);
+
+	ctx->dsp->ops->free(ctx->dsp);
+}
+EXPORT_SYMBOL_GPL(cnl_sst_dsp_cleanup);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Cannonlake IPC driver");
-- 
cgit v1.2.3


From b003a345dc67c7a1c468ad8ff10b24d7dd12e12c Mon Sep 17 00:00:00 2001
From: Guneshwor Singh
Date: Wed, 2 Aug 2017 21:51:19 +0530
Subject: ASoC: Intel: cnl: add dsp ops for cannonlake

Add cannonlake dsp support by adding its dsp_ops.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 9 +++++++++
 1 file changed, 9 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index c6b60ee5b196..bfb3332a77ca 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -22,6 +22,7 @@
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include "skl-sst-dsp.h"
+#include "cnl-sst-dsp.h"
 #include "skl-sst-ipc.h"
 #include "skl.h"
 #include "../common/sst-dsp.h"
@@ -231,6 +232,14 @@ static const struct skl_dsp_ops dsp_ops[] = {
 		.init_fw = bxt_sst_init_fw,
 		.cleanup = bxt_sst_dsp_cleanup
 	},
+	{
+		.id = 0x9dc8,
+		.num_cores = 4,
+		.loader_ops = bxt_get_loader_ops,
+		.init = cnl_sst_dsp_init,
+		.init_fw = cnl_sst_init_fw,
+		.cleanup = cnl_sst_dsp_cleanup
+	},
 };
 
 const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id)
-- 
cgit v1.2.3


From 86d7ce3dd717e09a442d35dae3f9c62c03e024d8 Mon Sep 17 00:00:00 2001
From: Guneshwor Singh
Date: Wed, 2 Aug 2017 21:51:20 +0530
Subject: ASoC: Intel: cnl: add pci id for cnl

Enable cnl by adding its pci id in skl_ids[].

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl.c | 11 +++++++++++
 1 file changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index ab651517e8b8..e612a4097275 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -996,6 +996,14 @@ static struct sst_acpi_mach sst_glk_devdata[] = {
 	},
 };
 
+static const struct sst_acpi_mach sst_cnl_devdata[] = {
+	{
+		.id = "INT34C2",
+		.drv_name = "cnl_rt274",
+		.fw_filename = "intel/dsp_fw_cnl.bin",
+	},
+};
+
 /* PCI IDs */
 static const struct pci_device_id skl_ids[] = {
 	/* Sunrise Point-LP */
@@ -1010,6 +1018,9 @@ static const struct pci_device_id skl_ids[] = {
 	/* GLK */
 	{ PCI_DEVICE(0x8086, 0x3198),
 		.driver_data = (unsigned long)&sst_glk_devdata},
+	/* CNL */
+	{ PCI_DEVICE(0x8086, 0x9dc8),
+		.driver_data = (unsigned long)&sst_cnl_devdata},
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, skl_ids);
-- 
cgit v1.2.3


From 67e8ee9b4b387f5c4d12ad9b926a78a9bf3dc275 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Thu, 3 Aug 2017 23:20:01 +0530
Subject: ASoC: Intel: constify pci_device_id.

pci_device_id are not supposed to change at runtime. All functions
working with pci_device_id provided by <linux/pci.h> work with
const pci_device_id. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/atom/sst/sst_pci.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c
index 3a0b3bf0af97..6906ee624cf6 100644
--- a/sound/soc/intel/atom/sst/sst_pci.c
+++ b/sound/soc/intel/atom/sst/sst_pci.c
@@ -181,7 +181,7 @@ static void intel_sst_remove(struct pci_dev *pci)
 }
 
 /* PCI Routines */
-static struct pci_device_id intel_sst_ids[] = {
+static const struct pci_device_id intel_sst_ids[] = {
 	{ PCI_VDEVICE(INTEL, SST_MRFLD_PCI_ID), 0},
 	{ 0, }
 };
-- 
cgit v1.2.3


From 8d7313bfb1bb7d71a6ebc503215e4fb4a8964741 Mon Sep 17 00:00:00 2001
From: Christophe Jaillet
Date: Mon, 7 Aug 2017 02:23:15 +0200
Subject: ASoC: mediatek: Fix an error checking code

Check the value returned by 'devm_clk_get()' instead of the clock
identifier which can never be an ERR code.

Fixes: d6f3710a56e1 ("ASoC: mediatek: add structure define and clock control for 2701")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c b/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c
index b815ecc6bbf6..affa7fb25dd9 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c
@@ -75,7 +75,7 @@ int mt2701_init_clock(struct mtk_base_afe *afe)
 
 	for (i = 0; i < MT2701_CLOCK_NUM; i++) {
 		afe_priv->clocks[i] = devm_clk_get(afe->dev, aud_clks[i]);
-		if (IS_ERR(aud_clks[i])) {
+		if (IS_ERR(afe_priv->clocks[i])) {
 			dev_warn(afe->dev, "%s devm_clk_get %s fail\n",
 				 __func__, aud_clks[i]);
 			return PTR_ERR(aud_clks[i]);
-- 
cgit v1.2.3


From b56bff43a965ffe03b4a994fd1a102059d272807 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Mon, 7 Aug 2017 18:39:32 +0800
Subject: ASoC: rt5514: reset dma_offset at hw_params

dma_offset needs reset every time otherwise for consecutive PCM
open the last dma_offset value will cause incorrect pointer position
be reported to user space.

Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514-spi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 6f513cc4c56c..640193d845be 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -151,7 +151,6 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data)
 	u8 buf[8];
 
 	rt5514_dsp->get_size = 0;
-	rt5514_dsp->dma_offset = 0;
 
 	/**
 	 * The address area x1800XXXX is the register address, and it cannot
@@ -205,6 +204,7 @@ static int rt5514_spi_hw_params(struct snd_pcm_substream *substream,
 	ret = snd_pcm_lib_alloc_vmalloc_buffer(substream,
 			params_buffer_bytes(hw_params));
 	rt5514_dsp->substream = substream;
+	rt5514_dsp->dma_offset = 0;
 	mutex_unlock(&rt5514_dsp->dma_lock);
 
 	return ret;
-- 
cgit v1.2.3


From 92ff5d085ecc8255600551055262bf98a8f3f1d0 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 3 Aug 2017 21:30:20 +0530
Subject: ASoC: sunxi: make snd_soc_codec_driver structures as const

Declare snd_soc_codec_driver structures as const as they are either
passed as an argument to the function snd_soc_register_codec or stored as
reference in field codec of type sun4i_codec_quirks. Both the fucntion
argument and the codec field are of type const, so declare the
structures with this property as const.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-codec.c | 6 +++---
 sound/soc/sunxi/sun8i-codec.c | 2 +-
 2 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 73d054f466c2..c668b8b08b38 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -762,7 +762,7 @@ static const struct snd_soc_dapm_route sun4i_codec_codec_dapm_routes[] = {
 	{ "Mic1", NULL, "VMIC" },
 };
 
-static struct snd_soc_codec_driver sun4i_codec_codec = {
+static const struct snd_soc_codec_driver sun4i_codec_codec = {
 	.component_driver = {
 		.controls		= sun4i_codec_controls,
 		.num_controls		= ARRAY_SIZE(sun4i_codec_controls),
@@ -1068,7 +1068,7 @@ static const struct snd_soc_dapm_route sun6i_codec_codec_dapm_routes[] = {
 	{ "Right ADC", NULL, "Right ADC Mixer" },
 };
 
-static struct snd_soc_codec_driver sun6i_codec_codec = {
+static const struct snd_soc_codec_driver sun6i_codec_codec = {
 	.component_driver = {
 		.controls		= sun6i_codec_codec_widgets,
 		.num_controls		= ARRAY_SIZE(sun6i_codec_codec_widgets),
@@ -1096,7 +1096,7 @@ static const struct snd_soc_dapm_widget sun8i_a23_codec_codec_widgets[] = {
 
 };
 
-static struct snd_soc_codec_driver sun8i_a23_codec_codec = {
+static const struct snd_soc_codec_driver sun8i_a23_codec_codec = {
 	.component_driver = {
 		.controls		= sun8i_a23_codec_codec_controls,
 		.num_controls		= ARRAY_SIZE(sun8i_a23_codec_codec_controls),
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 253ae0b5a4d4..abfb710df7cb 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -360,7 +360,7 @@ static struct snd_soc_dai_driver sun8i_codec_dai = {
 	.ops = &sun8i_codec_dai_ops,
 };
 
-static struct snd_soc_codec_driver sun8i_soc_codec = {
+static const struct snd_soc_codec_driver sun8i_soc_codec = {
 	.component_driver = {
 		.dapm_widgets		= sun8i_codec_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(sun8i_codec_dapm_widgets),
-- 
cgit v1.2.3


From 353c64dd19ebc99786c3578ea0444e1b5b1bb172 Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Mon, 7 Aug 2017 09:08:52 -0300
Subject: ASoC: soc-core: Remove unneeded dentry member from snd_soc_codec

There is no need to have the *debugfs_reg dentry member as part of
the snd_soc_codec structure as its only usage is inside
soc_init_codec_debugfs().

Use a local dentry variable instead.

Signed-off-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 4 ----
 sound/soc/soc-core.c | 9 +++++----
 2 files changed, 5 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 10b71beb9f13..aad3f14966f1 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -903,10 +903,6 @@ struct snd_soc_codec {
 
 	/* component */
 	struct snd_soc_component component;
-
-#ifdef CONFIG_DEBUG_FS
-	struct dentry *debugfs_reg;
-#endif
 };
 
 /* codec driver */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 921622a01944..558efdfd717d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -339,11 +339,12 @@ static void soc_cleanup_component_debugfs(struct snd_soc_component *component)
 static void soc_init_codec_debugfs(struct snd_soc_component *component)
 {
 	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
+	struct dentry *debugfs_reg;
 
-	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
-						 codec->component.debugfs_root,
-						 codec, &codec_reg_fops);
-	if (!codec->debugfs_reg)
+	debugfs_reg = debugfs_create_file("codec_reg", 0644,
+					  codec->component.debugfs_root,
+					  codec, &codec_reg_fops);
+	if (!debugfs_reg)
 		dev_warn(codec->dev,
 			"ASoC: Failed to create codec register debugfs file\n");
 }
-- 
cgit v1.2.3


From d9a02c552cd69c466998e3afdf420cb4c0b9d9d9 Mon Sep 17 00:00:00 2001
From: Fabio Estevam
Date: Mon, 7 Aug 2017 09:08:53 -0300
Subject: ASoC: soc-core: Use IS_ERR_OR_NULL()

The check of IS_ERR or NULL pointer can be replaced by IS_ERR_OR_NULL(),
which helps readability.

Signed-off-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 558efdfd717d..07d708498500 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -495,7 +495,7 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
 static void snd_soc_debugfs_init(void)
 {
 	snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL);
-	if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) {
+	if (IS_ERR_OR_NULL(snd_soc_debugfs_root)) {
 		pr_warn("ASoC: Failed to create debugfs directory\n");
 		snd_soc_debugfs_root = NULL;
 		return;
-- 
cgit v1.2.3


From cf9e829eaf0195e722bc3a3929002a3fbe74dafa Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Mon, 7 Aug 2017 02:06:23 +0000
Subject: ASoC: soc-core: rename "cmpnt" to "component"

To unify notation, to readable.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  4 ++--
 sound/soc/soc-core.c | 36 ++++++++++++++++++------------------
 2 files changed, 20 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9c94b97c17f8..de57a8ad057b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -469,10 +469,10 @@ int snd_soc_register_codec(struct device *dev,
 		struct snd_soc_dai_driver *dai_drv, int num_dai);
 void snd_soc_unregister_codec(struct device *dev);
 int snd_soc_register_component(struct device *dev,
-			 const struct snd_soc_component_driver *cmpnt_drv,
+			 const struct snd_soc_component_driver *component_driver,
 			 struct snd_soc_dai_driver *dai_drv, int num_dai);
 int devm_snd_soc_register_component(struct device *dev,
-			 const struct snd_soc_component_driver *cmpnt_drv,
+			 const struct snd_soc_component_driver *component_driver,
 			 struct snd_soc_dai_driver *dai_drv, int num_dai);
 void snd_soc_unregister_component(struct device *dev);
 int snd_soc_cache_init(struct snd_soc_codec *codec);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 921622a01944..b21488972198 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3282,40 +3282,40 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
 }
 
 int snd_soc_register_component(struct device *dev,
-			       const struct snd_soc_component_driver *cmpnt_drv,
+			       const struct snd_soc_component_driver *component_driver,
 			       struct snd_soc_dai_driver *dai_drv,
 			       int num_dai)
 {
-	struct snd_soc_component *cmpnt;
+	struct snd_soc_component *component;
 	int ret;
 
-	cmpnt = kzalloc(sizeof(*cmpnt), GFP_KERNEL);
-	if (!cmpnt) {
+	component = kzalloc(sizeof(*component), GFP_KERNEL);
+	if (!component) {
 		dev_err(dev, "ASoC: Failed to allocate memory\n");
 		return -ENOMEM;
 	}
 
-	ret = snd_soc_component_initialize(cmpnt, cmpnt_drv, dev);
+	ret = snd_soc_component_initialize(component, component_driver, dev);
 	if (ret)
 		goto err_free;
 
-	cmpnt->ignore_pmdown_time = true;
-	cmpnt->registered_as_component = true;
+	component->ignore_pmdown_time = true;
+	component->registered_as_component = true;
 
-	ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai, true);
+	ret = snd_soc_register_dais(component, dai_drv, num_dai, true);
 	if (ret < 0) {
 		dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret);
 		goto err_cleanup;
 	}
 
-	snd_soc_component_add(cmpnt);
+	snd_soc_component_add(component);
 
 	return 0;
 
 err_cleanup:
-	snd_soc_component_cleanup(cmpnt);
+	snd_soc_component_cleanup(component);
 err_free:
-	kfree(cmpnt);
+	kfree(component);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_register_component);
@@ -3327,22 +3327,22 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component);
  */
 void snd_soc_unregister_component(struct device *dev)
 {
-	struct snd_soc_component *cmpnt;
+	struct snd_soc_component *component;
 
 	mutex_lock(&client_mutex);
-	list_for_each_entry(cmpnt, &component_list, list) {
-		if (dev == cmpnt->dev && cmpnt->registered_as_component)
+	list_for_each_entry(component, &component_list, list) {
+		if (dev == component->dev && component->registered_as_component)
 			goto found;
 	}
 	mutex_unlock(&client_mutex);
 	return;
 
 found:
-	snd_soc_tplg_component_remove(cmpnt, SND_SOC_TPLG_INDEX_ALL);
-	snd_soc_component_del_unlocked(cmpnt);
+	snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
+	snd_soc_component_del_unlocked(component);
 	mutex_unlock(&client_mutex);
-	snd_soc_component_cleanup(cmpnt);
-	kfree(cmpnt);
+	snd_soc_component_cleanup(component);
+	kfree(component);
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
 
-- 
cgit v1.2.3


From 21a035287e28ec225fe7dfd1fffc509efd26a5db Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Mon, 7 Aug 2017 02:06:40 +0000
Subject: ASoC: soc-core: remove duplicate mutex_unlock from
 snd_soc_unregister_component()

Current snd_soc_unregister_component() is using multiple mutex_unlock()
for found/non-found cases. But it is unreadable and confusable code.
This patch tidyup current code to be readable.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 22 +++++++++++++---------
 1 file changed, 13 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b21488972198..3bb8c63564cf 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3328,21 +3328,25 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component);
 void snd_soc_unregister_component(struct device *dev)
 {
 	struct snd_soc_component *component;
+	int found = 0;
 
 	mutex_lock(&client_mutex);
 	list_for_each_entry(component, &component_list, list) {
-		if (dev == component->dev && component->registered_as_component)
-			goto found;
+		if (dev != component->dev ||
+		    !component->registered_as_component)
+			continue;
+
+		snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
+		snd_soc_component_del_unlocked(component);
+		found = 1;
+		break;
 	}
 	mutex_unlock(&client_mutex);
-	return;
 
-found:
-	snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
-	snd_soc_component_del_unlocked(component);
-	mutex_unlock(&client_mutex);
-	snd_soc_component_cleanup(component);
-	kfree(component);
+	if (found) {
+		snd_soc_component_cleanup(component);
+		kfree(component);
+	}
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
 
-- 
cgit v1.2.3


From 06d15a2ede999a77f0d05aa600ed473d90d9a909 Mon Sep 17 00:00:00 2001
From: Rob Herring
Date: Mon, 7 Aug 2017 18:29:18 -0500
Subject: ASoC: fsl: Convert to using %pOF instead of full_name

Now that we have a custom printf format specifier, convert users of
full_name to use %pOF instead. This is preparation to remove storing
of the full path string for each node.

Signed-off-by: Rob Herring <robh@kernel.org>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_dma.c    |  4 ++--
 sound/soc/fsl/imx-audmux.c | 16 ++++++++--------
 2 files changed, 10 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index ccadefceeff2..ed8ea002902d 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -897,8 +897,8 @@ static int fsl_soc_dma_probe(struct platform_device *pdev)
 
 	ret = of_address_to_resource(ssi_np, 0, &res);
 	if (ret) {
-		dev_err(&pdev->dev, "could not determine resources for %s\n",
-			ssi_np->full_name);
+		dev_err(&pdev->dev, "could not determine resources for %pOF\n",
+			ssi_np);
 		of_node_put(ssi_np);
 		return ret;
 	}
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index fc57da341d61..392d5eef356d 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -268,13 +268,13 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev,
 
 		ret = of_property_read_u32(child, "fsl,audmux-port", &port);
 		if (ret) {
-			dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n",
-					child->full_name);
+			dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%pOF\"\n",
+					child);
 			continue;
 		}
 		if (!of_property_read_bool(child, "fsl,port-config")) {
-			dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n",
-					child->full_name);
+			dev_warn(&pdev->dev, "child node \"%pOF\" does not have property fsl,port-config\n",
+					child);
 			continue;
 		}
 
@@ -292,15 +292,15 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev,
 		}
 
 		if (ret != -EOVERFLOW) {
-			dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n",
-					i, child->full_name);
+			dev_err(&pdev->dev, "Failed to read u32 at index %d of child %pOF\n",
+					i, child);
 			continue;
 		}
 
 		if (audmux_type == IMX31_AUDMUX) {
 			if (i % 2) {
-				dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n",
-						child->full_name);
+				dev_err(&pdev->dev, "One pdcr value is missing in child node %pOF\n",
+						child);
 				continue;
 			}
 			imx_audmux_v2_configure_port(port, ptcr, pdcr);
-- 
cgit v1.2.3


From a8ad0c85fbfa56c44161829f6433e59dcd84b731 Mon Sep 17 00:00:00 2001
From: Sylwester Nawrocki
Date: Fri, 4 Aug 2017 12:58:17 +0200
Subject: ASoC: samsung: odroid: Drop requirement of clocks in the sound node

As suggested in reviews the requirement of clocks in the 'sound' node
is dropped and instead a leaf clock is used to configure frequency
of the audio root clock PLL. This can work now after the clock tree
definitions have been updated to allow clock rate setting propagation
on the path from the I2S controller up to the EPLL.

This patch also lowers the CODEC master clock frequency so as
to not exceed the maximum allowed 60 MHz at maximum audio sampling
rates.

Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/samsung,odroid.txt   |  6 ---
 sound/soc/samsung/odroid.c                         | 44 +++++++++++++++-------
 2 files changed, 30 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt
index c30934dd975b..625b1b18fd02 100644
--- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt
+++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt
@@ -7,9 +7,6 @@ Required properties:
  - model - the user-visible name of this sound complex
  - clocks - should contain entries matching clock names in the clock-names
     property
- - clock-names - should contain following entries:
-    - "epll" - indicating the EPLL output clock
-    - "i2s_rclk" - indicating the RCLK (root) clock of the I2S0 controller
  - samsung,audio-widgets - this property specifies off-codec audio elements
    like headphones or speakers, for details see widgets.txt
  - samsung,audio-routing - a list of the connections between audio
@@ -46,9 +43,6 @@ sound {
 		"IN1", "Mic Jack",
 		"Mic Jack", "MICBIAS";
 
-	clocks = <&clock CLK_FOUT_EPLL>, <&i2s0 CLK_I2S_RCLK_SRC>;
-	clock-names = "epll", "sclk_i2s";
-
 	cpu {
 		sound-dai = <&i2s0 0>;
 	};
diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index 0c0b00e40646..00b7f455a472 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -19,8 +19,8 @@ struct odroid_priv {
 	struct snd_soc_card card;
 	struct snd_soc_dai_link dai_link;
 
-	struct clk *pll;
-	struct clk *rclk;
+	struct clk *clk_i2s_bus;
+	struct clk *sclk_i2s;
 };
 
 static int odroid_card_startup(struct snd_pcm_substream *substream)
@@ -58,13 +58,18 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	ret = clk_set_rate(priv->pll, pll_freq + 1);
+	ret = clk_set_rate(priv->clk_i2s_bus, pll_freq / 2 + 1);
 	if (ret < 0)
 		return ret;
 
-	rclk_freq = params_rate(params) * 256 * 4;
+	/*
+	 *  We add 1 to the rclk_freq value in order to avoid too low clock
+	 *  frequency values due to the EPLL output frequency not being exact
+	 *  multiple of the audio sampling rate.
+	 */
+	rclk_freq = params_rate(params) * 256 + 1;
 
-	ret = clk_set_rate(priv->rclk, rclk_freq);
+	ret = clk_set_rate(priv->sclk_i2s, rclk_freq);
 	if (ret < 0)
 		return ret;
 
@@ -118,14 +123,6 @@ static int odroid_audio_probe(struct platform_device *pdev)
 
 	snd_soc_card_set_drvdata(card, priv);
 
-	priv->pll = devm_clk_get(dev, "epll");
-	if (IS_ERR(priv->pll))
-		return PTR_ERR(priv->pll);
-
-	priv->rclk = devm_clk_get(dev, "i2s_rclk");
-	if (IS_ERR(priv->rclk))
-		return PTR_ERR(priv->rclk);
-
 	ret = snd_soc_of_parse_card_name(card, "model");
 	if (ret < 0)
 		return ret;
@@ -171,14 +168,31 @@ static int odroid_audio_probe(struct platform_device *pdev)
 	link->name = "Primary";
 	link->stream_name = link->name;
 
+
+	priv->sclk_i2s = of_clk_get_by_name(link->cpu_of_node, "i2s_opclk1");
+	if (IS_ERR(priv->sclk_i2s)) {
+		ret = PTR_ERR(priv->sclk_i2s);
+		goto err_put_i2s_n;
+	}
+
+	priv->clk_i2s_bus = of_clk_get_by_name(link->cpu_of_node, "iis");
+	if (IS_ERR(priv->clk_i2s_bus)) {
+		ret = PTR_ERR(priv->clk_i2s_bus);
+		goto err_put_sclk;
+	}
+
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
 		dev_err(dev, "snd_soc_register_card() failed: %d\n", ret);
-		goto err_put_i2s_n;
+		goto err_put_clk_i2s;
 	}
 
 	return 0;
 
+err_put_clk_i2s:
+	clk_put(priv->clk_i2s_bus);
+err_put_sclk:
+	clk_put(priv->sclk_i2s);
 err_put_i2s_n:
 	of_node_put(link->cpu_of_node);
 err_put_codec_n:
@@ -192,6 +206,8 @@ static int odroid_audio_remove(struct platform_device *pdev)
 
 	of_node_put(priv->dai_link.cpu_of_node);
 	odroid_put_codec_of_nodes(&priv->dai_link);
+	clk_put(priv->sclk_i2s);
+	clk_put(priv->clk_i2s_bus);
 
 	return 0;
 }
-- 
cgit v1.2.3


From bf23c6a3813aac96e0af01565aba012b25c8bae5 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 8 Aug 2017 01:32:08 +0000
Subject: ASoC: rsnd: move rsnd_ssi_config_init() execute condition into it.

Make it the same style as other functions

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 49cdc5e007b3..3ee7b2de7dea 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -335,6 +335,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	u32 wsr;
 	int is_tdm;
 
+	if (rsnd_ssi_is_parent(mod, io))
+		return;
+
 	is_tdm = rsnd_runtime_is_ssi_tdm(io);
 
 	/*
@@ -472,8 +475,7 @@ static int rsnd_ssi_init(struct rsnd_mod *mod,
 	if (ret < 0)
 		return ret;
 
-	if (!rsnd_ssi_is_parent(mod, io))
-		rsnd_ssi_config_init(mod, io);
+	rsnd_ssi_config_init(mod, io);
 
 	rsnd_ssi_register_setup(mod);
 
-- 
cgit v1.2.3


From bb002f925ae18c0c9d72d87b74b311a5209d7178 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 8 Aug 2017 01:32:42 +0000
Subject: ASoC: rsnd: rsnd_ssi_can_output_clk() macro

For example SSI0/SSI1 case, SSI1 can share pin with SSI0.
And then, SSI1 needs SSI0's clock.
This clock controlling is very picky and difficult to understand
in current code.

This patch adds new rsnd_ssi_can_output_clk() macro,
the code will be more understandable.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 3ee7b2de7dea..434996d4054a 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -106,6 +106,7 @@ struct rsnd_ssi {
 	(rsnd_ssi_multi_slaves(io) & (1 << rsnd_mod_id(mod)))
 #define rsnd_ssi_is_run_mods(mod, io) \
 	(rsnd_ssi_run_mods(io) & (1 << rsnd_mod_id(mod)))
+#define rsnd_ssi_can_output_clk(mod) (!__rsnd_ssi_is_pin_sharing(mod))
 
 int rsnd_ssi_hdmi_port(struct rsnd_dai_stream *io)
 {
@@ -255,7 +256,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io);
 	int chan = rsnd_runtime_channel_for_ssi(io);
 	int idx, ret;
 	unsigned int main_rate;
@@ -266,7 +266,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	if (!rsnd_rdai_is_clk_master(rdai))
 		return 0;
 
-	if (ssi_parent_mod && !rsnd_ssi_is_parent(mod, io))
+	if (!rsnd_ssi_can_output_clk(mod))
 		return 0;
 
 	if (rsnd_ssi_is_multi_slave(mod, io))
@@ -307,12 +307,11 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_mod *mod,
 {
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io);
 
 	if (!rsnd_rdai_is_clk_master(rdai))
 		return;
 
-	if (ssi_parent_mod && !rsnd_ssi_is_parent(mod, io))
+	if (!rsnd_ssi_can_output_clk(mod))
 		return;
 
 	if (ssi->usrcnt > 1)
-- 
cgit v1.2.3


From 597b046f0d99b0b0123a715f8c8b1ea881d2bec6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 8 Aug 2017 01:31:39 +0000
Subject: ASoC: rsnd: control SSICR::EN correctly

In case of SSI0 playback, SSI1 capture, SSI0 might be shared for
clock output if clock master mode.

Current rsnd driver had been assumed that SSI clock contiguous
output which is needed for SSI parent needs SSICR::EN (SSI module
enable) bit.
But, this bit controls data input/output, not for clock.
Clock contiguous output needs SSICR : FORCE, SCKD, SWSD,
and SSIWSR : CONT. Not SSICR : EN.

Because of this wrong assumption, and insufficient control, on current
code, for example, if it starts SSI0(playback) -> SSI1(capture) order,
SSI0 SSICR::EN bit will temporarily be 0.
It causes playback side underrun error. This is bug.
We can reproduce this issue with SSI+SRC (without DVC), and capture
during playback operation.

This patch fixup current (wrong) assumption, and control SSICR::EN bit
correctly.

Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 40 +++++++++++++++++++++++++++++++---------
 1 file changed, 31 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 46feddd78ee2..c5d5c64152e9 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -72,6 +72,7 @@ struct rsnd_ssi {
 	u32 cr_own;
 	u32 cr_clk;
 	u32 cr_mode;
+	u32 cr_en;
 	u32 wsr;
 	int chan;
 	int rate;
@@ -291,6 +292,16 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	if (ret < 0)
 		return ret;
 
+	/*
+	 * SSI clock will be output contiguously
+	 * by below settings.
+	 * This means, rsnd_ssi_master_clk_start()
+	 * and rsnd_ssi_register_setup() are necessary
+	 * for SSI parent
+	 *
+	 * SSICR  : FORCE, SCKD, SWSD
+	 * SSIWSR : CONT
+	 */
 	ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(idx);
 	ssi->wsr = CONT;
 	ssi->rate = rate;
@@ -393,7 +404,8 @@ static void rsnd_ssi_register_setup(struct rsnd_mod *mod)
 	rsnd_mod_write(mod, SSIWSR,	ssi->wsr);
 	rsnd_mod_write(mod, SSICR,	ssi->cr_own	|
 					ssi->cr_clk	|
-					ssi->cr_mode); /* without EN */
+					ssi->cr_mode	|
+					ssi->cr_en);
 }
 
 static void rsnd_ssi_pointer_init(struct rsnd_mod *mod,
@@ -544,6 +556,8 @@ static int rsnd_ssi_start(struct rsnd_mod *mod,
 			  struct rsnd_dai_stream *io,
 			  struct rsnd_priv *priv)
 {
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+
 	if (!rsnd_ssi_is_run_mods(mod, io))
 		return 0;
 
@@ -554,7 +568,19 @@ static int rsnd_ssi_start(struct rsnd_mod *mod,
 	if (rsnd_ssi_multi_slaves_runtime(io))
 		return 0;
 
-	rsnd_mod_bset(mod, SSICR, EN, EN);
+	/*
+	 * EN is for data output.
+	 * SSI parent EN is not needed.
+	 */
+	if (rsnd_ssi_is_parent(mod, io))
+		return 0;
+
+	ssi->cr_en = EN;
+
+	rsnd_mod_write(mod, SSICR,	ssi->cr_own	|
+					ssi->cr_clk	|
+					ssi->cr_mode	|
+					ssi->cr_en);
 
 	return 0;
 }
@@ -569,13 +595,7 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod,
 	if (!rsnd_ssi_is_run_mods(mod, io))
 		return 0;
 
-	/*
-	 * don't stop if not last user
-	 * see also
-	 *	rsnd_ssi_start
-	 *	rsnd_ssi_interrupt
-	 */
-	if (ssi->usrcnt > 1)
+	if (rsnd_ssi_is_parent(mod, io))
 		return 0;
 
 	/*
@@ -595,6 +615,8 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod,
 	rsnd_mod_write(mod, SSICR, cr);	/* disabled all */
 	rsnd_ssi_status_check(mod, IIRQ);
 
+	ssi->cr_en = 0;
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From ade82fdbae6905e80f107596f2e0b4df84490d28 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Wed, 9 Aug 2017 09:56:14 +0800
Subject: ASoC: rt5514: Eliminate the noise in the ASRC case

In the ASRC case, there is the noise if the clock is asynchronous with the
LRCK. The default settings will eliminate the noise.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 21cbb566c0d3..0eabe569ab73 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -64,6 +64,8 @@ static const struct reg_sequence rt5514_patch[] = {
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_ADCFED,	0x00000800},
 	{RT5514_ASRC_IN_CTRL1,		0x00000003},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
 };
 
 static const struct reg_default rt5514_reg[] = {
@@ -89,10 +91,10 @@ static const struct reg_default rt5514_reg[] = {
 	{RT5514_DELAY_BUF_CTRL3,	0x00000000},
 	{RT5514_DOWNFILTER0_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER0_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER0_CTRL3,	0x00000362},
+	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
 	{RT5514_DOWNFILTER1_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER1_CTRL2,	0x00020c2f},
-	{RT5514_DOWNFILTER1_CTRL3,	0x00000362},
+	{RT5514_DOWNFILTER1_CTRL3,	0x10000362},
 	{RT5514_ANA_CTRL_LDO10,		0x00028604},
 	{RT5514_ANA_CTRL_LDO18_16,	0x02000345},
 	{RT5514_ANA_CTRL_ADC12,		0x0000a2a8},
-- 
cgit v1.2.3


From 29d03ff5c752e02918e8319fdcc86faa374d36ec Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Wed, 9 Aug 2017 01:46:23 +0000
Subject: ASoC: rsnd: remove 16ch support for now

R-Car Gen3 sound device can use TDM 16ch, but its usage is very
limited. It is not yet supported.
Let's remove unexpected/unsupported channel settings for now.
This patch makes xx_ADINR register settings sanity without
runtime check.

R-Car Gen3 datasheet

40.4.7 Note on Usage of the SSIU Module
When an SSIU module is used in "TDM-16ch mode",
only the transfer with the external memory is available.

Reported-by: Nguyen Viet Dung <dung.nguyen.aj@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/core.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 3f2ced26ed37..7fa40fcad59e 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -726,7 +726,6 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	case 2:
 	case 6:
 	case 8:
-	case 16:
 		/* TDM Extend Mode */
 		rsnd_rdai_channels_set(rdai, slots);
 		rsnd_rdai_ssi_lane_set(rdai, 1);
@@ -740,7 +739,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
 }
 
 static unsigned int rsnd_soc_hw_channels_list[] = {
-	2, 6, 8, 16,
+	2, 6, 8,
 };
 
 static unsigned int rsnd_soc_hw_rate_list[] = {
@@ -1017,7 +1016,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 	drv->playback.rates		= RSND_RATES;
 	drv->playback.formats		= RSND_FMTS;
 	drv->playback.channels_min	= 2;
-	drv->playback.channels_max	= 16;
+	drv->playback.channels_max	= 8;
 	drv->playback.stream_name	= rdai->playback.name;
 
 	snprintf(rdai->capture.name, RSND_DAI_NAME_SIZE,
@@ -1025,7 +1024,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 	drv->capture.rates		= RSND_RATES;
 	drv->capture.formats		= RSND_FMTS;
 	drv->capture.channels_min	= 2;
-	drv->capture.channels_max	= 16;
+	drv->capture.channels_max	= 8;
 	drv->capture.stream_name	= rdai->capture.name;
 
 	rdai->playback.rdai		= rdai;
-- 
cgit v1.2.3


From e0936c3471a8411a5df327641fa3ffe12a2fb07b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Wed, 9 Aug 2017 02:16:20 +0000
Subject: ASoC: rsnd: avoid duplicate free_irq()

commit 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on
Parent SSI") fixed Parent SSI duplicate free_irq().
But on Renesas Sound, not only Parent SSI but also Multi SSI
have same issue.
This patch avoid duplicate free_irq() if it was not pure SSI.

Fixes: 1f8754d4daea5f ("ASoC: rsnd: don't call free_irq() on Parent SSI")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index c5d5c64152e9..90cc9c13c4e0 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -845,10 +845,10 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod,
 			       struct rsnd_priv *priv)
 {
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io);
+	struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io);
 
-	/* Do nothing for SSI parent mod */
-	if (ssi_parent_mod == mod)
+	/* Do nothing if non SSI (= SSI parent, multi SSI) mod */
+	if (pure_ssi_mod != mod)
 		return 0;
 
 	/* PIO will request IRQ again */
-- 
cgit v1.2.3


From 65d55e8a3bb2d2a98078590cfa49cd508aa80e81 Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Thu, 10 Aug 2017 09:50:51 +0200
Subject: ASoC: spear: Delete an error message for a failed memory allocation
 in two functions

Omit an extra message for a memory allocation failure in these functions.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/spear/spdif_in.c  | 4 +---
 sound/soc/spear/spdif_out.c | 4 +---
 2 files changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 7f32527fc3c8..01b28e59b4c0 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -216,10 +216,8 @@ static int spdif_in_probe(struct platform_device *pdev)
 		return -EINVAL;
 
 	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
-	if (!host) {
-		dev_warn(&pdev->dev, "kzalloc fail\n");
+	if (!host)
 		return -ENOMEM;
-	}
 
 	host->io_base = io_base;
 	host->irq = platform_get_irq(pdev, 0);
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index 0a72d52d533e..58d5843811f9 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -282,10 +282,8 @@ static int spdif_out_probe(struct platform_device *pdev)
 	int ret;
 
 	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
-	if (!host) {
-		dev_warn(&pdev->dev, "kzalloc fail\n");
+	if (!host)
 		return -ENOMEM;
-	}
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	host->io_base = devm_ioremap_resource(&pdev->dev, res);
-- 
cgit v1.2.3


From 213691c7e873206742b39751895d15b41a02b3d1 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 10 Aug 2017 00:07:00 +0000
Subject: ASoC: rsnd: call free_irq() both DMA/PIO mode

commit 701172dca15b ("ASoC: rsnd: don't use devm_request_irq() for SSI")
exchanged devm_request_irq() to request_irq() for SSI, because SSI will
fallback into PIO mode if DMA doesn't work.
But, because of it, PIO mode needed to call free_irq() when removing,
not only DMA mode.
This patch call free_irq() both PIO/DMA. Otherwise, rsnd IRQ handler
will be increased if user tried many unbind/bind.

Fixes: 701172dca15b ("ASoC: rsnd: don't use devm_request_irq() for SSI")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 37 +++++++++++++++++++------------------
 1 file changed, 19 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 0309de58bf18..281ed0da6ba1 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -792,6 +792,23 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod,
 	return ret;
 }
 
+static int rsnd_ssi_common_remove(struct rsnd_mod *mod,
+				  struct rsnd_dai_stream *io,
+				  struct rsnd_priv *priv)
+{
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io);
+
+	/* Do nothing if non SSI (= SSI parent, multi SSI) mod */
+	if (pure_ssi_mod != mod)
+		return 0;
+
+	/* PIO will request IRQ again */
+	free_irq(ssi->irq, mod);
+
+	return 0;
+}
+
 static int rsnd_ssi_pointer(struct rsnd_mod *mod,
 			    struct rsnd_dai_stream *io,
 			    snd_pcm_uframes_t *pointer)
@@ -807,6 +824,7 @@ static int rsnd_ssi_pointer(struct rsnd_mod *mod,
 static struct rsnd_mod_ops rsnd_ssi_pio_ops = {
 	.name	= SSI_NAME,
 	.probe	= rsnd_ssi_common_probe,
+	.remove	= rsnd_ssi_common_remove,
 	.init	= rsnd_ssi_init,
 	.quit	= rsnd_ssi_quit,
 	.start	= rsnd_ssi_start,
@@ -841,23 +859,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
 	return ret;
 }
 
-static int rsnd_ssi_dma_remove(struct rsnd_mod *mod,
-			       struct rsnd_dai_stream *io,
-			       struct rsnd_priv *priv)
-{
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io);
-
-	/* Do nothing if non SSI (= SSI parent, multi SSI) mod */
-	if (pure_ssi_mod != mod)
-		return 0;
-
-	/* PIO will request IRQ again */
-	free_irq(ssi->irq, mod);
-
-	return 0;
-}
-
 static int rsnd_ssi_fallback(struct rsnd_mod *mod,
 			     struct rsnd_dai_stream *io,
 			     struct rsnd_priv *priv)
@@ -899,7 +900,7 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
 	.name	= SSI_NAME,
 	.dma_req = rsnd_ssi_dma_req,
 	.probe	= rsnd_ssi_dma_probe,
-	.remove	= rsnd_ssi_dma_remove,
+	.remove	= rsnd_ssi_common_remove,
 	.init	= rsnd_ssi_init,
 	.quit	= rsnd_ssi_quit,
 	.start	= rsnd_ssi_start,
-- 
cgit v1.2.3


From bf9b29c784b393d1e506945f1e24e871d2587eb3 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 10 Aug 2017 00:07:20 +0000
Subject: ASoC: rsnd: add rsnd_ssi_flags_has/add() macro

Current rsnd_ssi_mode_flags() is not useful. This patch adds
new rsnd_ssi_flags_has/add() macro for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 19 ++++++++++---------
 1 file changed, 10 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 281ed0da6ba1..ca9355b86bd9 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -101,7 +101,8 @@ struct rsnd_ssi {
 #define rsnd_ssi_to_dma(mod) ((ssi)->dma)
 #define rsnd_ssi_nr(priv) ((priv)->ssi_nr)
 #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
-#define rsnd_ssi_mode_flags(p) ((p)->flags)
+#define rsnd_ssi_flags_has(p, f) ((p)->flags & f)
+#define rsnd_ssi_flags_set(p, f) ((p)->flags |= f)
 #define rsnd_ssi_is_parent(ssi, io) ((ssi) == rsnd_io_to_mod_ssip(io))
 #define rsnd_ssi_is_multi_slave(mod, io) \
 	(rsnd_ssi_multi_slaves(io) & (1 << rsnd_mod_id(mod)))
@@ -114,10 +115,10 @@ int rsnd_ssi_hdmi_port(struct rsnd_dai_stream *io)
 	struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io);
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 
-	if (rsnd_ssi_mode_flags(ssi) & RSND_SSI_HDMI0)
+	if (rsnd_ssi_flags_has(ssi, RSND_SSI_HDMI0))
 		return RSND_SSI_HDMI_PORT0;
 
-	if (rsnd_ssi_mode_flags(ssi) & RSND_SSI_HDMI1)
+	if (rsnd_ssi_flags_has(ssi, RSND_SSI_HDMI1))
 		return RSND_SSI_HDMI_PORT1;
 
 	return 0;
@@ -132,7 +133,7 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io)
 	if (!rsnd_ssi_is_dma_mode(mod))
 		return 0;
 
-	if (!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_NO_BUSIF))
+	if (!(rsnd_ssi_flags_has(ssi, RSND_SSI_NO_BUSIF)))
 		use_busif = 1;
 	if (rsnd_io_to_mod_src(io))
 		use_busif = 1;
@@ -986,13 +987,13 @@ static void __rsnd_ssi_parse_hdmi_connection(struct rsnd_priv *priv,
 	ssi  = rsnd_mod_to_ssi(mod);
 
 	if (strstr(remote_ep->full_name, "hdmi0")) {
-		ssi->flags |= RSND_SSI_HDMI0;
+		rsnd_ssi_flags_set(ssi, RSND_SSI_HDMI0);
 		dev_dbg(dev, "%s[%d] connected to HDMI0\n",
 			 rsnd_mod_name(mod), rsnd_mod_id(mod));
 	}
 
 	if (strstr(remote_ep->full_name, "hdmi1")) {
-		ssi->flags |= RSND_SSI_HDMI1;
+		rsnd_ssi_flags_set(ssi, RSND_SSI_HDMI1);
 		dev_dbg(dev, "%s[%d] connected to HDMI1\n",
 			rsnd_mod_name(mod), rsnd_mod_id(mod));
 	}
@@ -1025,7 +1026,7 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
 {
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 
-	return !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_CLK_PIN_SHARE);
+	return !!(rsnd_ssi_flags_has(ssi, RSND_SSI_CLK_PIN_SHARE));
 }
 
 static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io,
@@ -1108,10 +1109,10 @@ int rsnd_ssi_probe(struct rsnd_priv *priv)
 		}
 
 		if (of_get_property(np, "shared-pin", NULL))
-			ssi->flags |= RSND_SSI_CLK_PIN_SHARE;
+			rsnd_ssi_flags_set(ssi, RSND_SSI_CLK_PIN_SHARE);
 
 		if (of_get_property(np, "no-busif", NULL))
-			ssi->flags |= RSND_SSI_NO_BUSIF;
+			rsnd_ssi_flags_set(ssi, RSND_SSI_NO_BUSIF);
 
 		ssi->irq = irq_of_parse_and_map(np, 0);
 		if (!ssi->irq) {
-- 
cgit v1.2.3


From b6e58fcacb3edc6245ecf700207ba675cba18018 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 10 Aug 2017 00:07:38 +0000
Subject: ASoC: rsnd: call request_irq/free_irq once in MIX case

Each module's dai callback function availability is controlled
by mod->status. For example "always called", "call once".
In .probe/.remove case, it needs to be called always, because
.probe will call xxx_attach() function on .probe, especially
if platform is using MIXer.
For example, below case, MIX0/DVC0/SSI0 needs to be called twice.

        playback = <&src0 &ctu02 &mix0 &dvc0 &ssi0>;
        playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>;

But in this case, SSI0 will call request_irq() twice.
This patch add new RSND_SSI_PROBED flag and control it

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/ssi.c | 27 ++++++++++++++++++++++-----
 1 file changed, 22 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index ca9355b86bd9..8bb47144660b 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -90,6 +90,7 @@ struct rsnd_ssi {
 #define RSND_SSI_NO_BUSIF		(1 << 1) /* SSI+DMA without BUSIF */
 #define RSND_SSI_HDMI0			(1 << 2) /* for HDMI0 */
 #define RSND_SSI_HDMI1			(1 << 3) /* for HDMI1 */
+#define RSND_SSI_PROBED			(1 << 4)
 
 #define for_each_rsnd_ssi(pos, priv, i)					\
 	for (i = 0;							\
@@ -103,6 +104,7 @@ struct rsnd_ssi {
 #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
 #define rsnd_ssi_flags_has(p, f) ((p)->flags & f)
 #define rsnd_ssi_flags_set(p, f) ((p)->flags |= f)
+#define rsnd_ssi_flags_del(p, f) ((p)->flags = ((p)->flags & ~f))
 #define rsnd_ssi_is_parent(ssi, io) ((ssi) == rsnd_io_to_mod_ssip(io))
 #define rsnd_ssi_is_multi_slave(mod, io) \
 	(rsnd_ssi_multi_slaves(io) & (1 << rsnd_mod_id(mod)))
@@ -784,11 +786,22 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod,
 	/*
 	 * SSI might be called again as PIO fallback
 	 * It is easy to manual handling for IRQ request/free
+	 *
+	 * OTOH, this function might be called many times if platform is
+	 * using MIX. It needs xxx_attach() many times on xxx_probe().
+	 * Because of it, we can't control .probe/.remove calling count by
+	 * mod->status.
+	 * But it don't need to call request_irq() many times.
+	 * Let's control it by RSND_SSI_PROBED flag.
 	 */
-	ret = request_irq(ssi->irq,
-			  rsnd_ssi_interrupt,
-			  IRQF_SHARED,
-			  dev_name(dev), mod);
+	if (!rsnd_ssi_flags_has(ssi, RSND_SSI_PROBED)) {
+		ret = request_irq(ssi->irq,
+				  rsnd_ssi_interrupt,
+				  IRQF_SHARED,
+				  dev_name(dev), mod);
+
+		rsnd_ssi_flags_set(ssi, RSND_SSI_PROBED);
+	}
 
 	return ret;
 }
@@ -805,7 +818,11 @@ static int rsnd_ssi_common_remove(struct rsnd_mod *mod,
 		return 0;
 
 	/* PIO will request IRQ again */
-	free_irq(ssi->irq, mod);
+	if (rsnd_ssi_flags_has(ssi, RSND_SSI_PROBED)) {
+		free_irq(ssi->irq, mod);
+
+		rsnd_ssi_flags_del(ssi, RSND_SSI_PROBED);
+	}
 
 	return 0;
 }
-- 
cgit v1.2.3


From a180ba45b1cf630b3bd5912ce235b2ee16606b8e Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 3 Aug 2017 21:30:19 +0530
Subject: ASoC: codecs: add const to snd_soc_codec_driver structures

Declare snd_soc_codec_driver structures as const as they are only passed
as an argument to the function snd_soc_register_codec. This argument is
of type const, so declare the structures with this property as const.
In file codecs/sn95031.c, snd_soc_codec_driver structure is also used in
a copy operation along with getting passed to snd_soc_register_codec.
So, it can be made const too.
Done using Coccinelle:

@match disable optional_qualifier@
identifier s;
position p;
@@
static struct snd_soc_codec_driver s@p={...};

@good1@
identifier match.s;
position p;
@@
snd_soc_register_codec(...,&s@p,...)

@bad@
identifier match.s;
position p!={match.p,good1.p};
@@
s@p

@depends on !bad disable optional_qualifier@
identifier match.s;
@@
static
+const
struct snd_soc_codec_driver s={...};

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/88pm860x-codec.c      | 2 +-
 sound/soc/codecs/ab8500-codec.c        | 2 +-
 sound/soc/codecs/ac97.c                | 2 +-
 sound/soc/codecs/ad1836.c              | 2 +-
 sound/soc/codecs/ad193x.c              | 2 +-
 sound/soc/codecs/ad1980.c              | 2 +-
 sound/soc/codecs/ad73311.c             | 2 +-
 sound/soc/codecs/adau1373.c            | 2 +-
 sound/soc/codecs/adau1701.c            | 2 +-
 sound/soc/codecs/adau1977.c            | 2 +-
 sound/soc/codecs/adav80x.c             | 2 +-
 sound/soc/codecs/ads117x.c             | 2 +-
 sound/soc/codecs/ak4104.c              | 2 +-
 sound/soc/codecs/ak4535.c              | 2 +-
 sound/soc/codecs/ak4554.c              | 2 +-
 sound/soc/codecs/ak4613.c              | 2 +-
 sound/soc/codecs/ak4641.c              | 2 +-
 sound/soc/codecs/ak4642.c              | 2 +-
 sound/soc/codecs/ak4671.c              | 2 +-
 sound/soc/codecs/ak5386.c              | 2 +-
 sound/soc/codecs/alc5623.c             | 2 +-
 sound/soc/codecs/bt-sco.c              | 2 +-
 sound/soc/codecs/cq93vc.c              | 2 +-
 sound/soc/codecs/cs35l33.c             | 2 +-
 sound/soc/codecs/cs35l34.c             | 2 +-
 sound/soc/codecs/cs35l35.c             | 2 +-
 sound/soc/codecs/cs4271.c              | 2 +-
 sound/soc/codecs/cs42l51.c             | 2 +-
 sound/soc/codecs/cs4349.c              | 2 +-
 sound/soc/codecs/cs47l24.c             | 2 +-
 sound/soc/codecs/cs53l30.c             | 2 +-
 sound/soc/codecs/cx20442.c             | 2 +-
 sound/soc/codecs/da7210.c              | 2 +-
 sound/soc/codecs/da7213.c              | 2 +-
 sound/soc/codecs/da7218.c              | 2 +-
 sound/soc/codecs/da7219.c              | 2 +-
 sound/soc/codecs/da732x.c              | 2 +-
 sound/soc/codecs/da9055.c              | 2 +-
 sound/soc/codecs/dmic.c                | 2 +-
 sound/soc/codecs/es7134.c              | 2 +-
 sound/soc/codecs/es8316.c              | 2 +-
 sound/soc/codecs/es8328.c              | 2 +-
 sound/soc/codecs/hdac_hdmi.c           | 2 +-
 sound/soc/codecs/hdmi-codec.c          | 2 +-
 sound/soc/codecs/ics43432.c            | 2 +-
 sound/soc/codecs/inno_rk3036.c         | 2 +-
 sound/soc/codecs/isabelle.c            | 2 +-
 sound/soc/codecs/jz4740.c              | 2 +-
 sound/soc/codecs/lm49453.c             | 2 +-
 sound/soc/codecs/max98088.c            | 2 +-
 sound/soc/codecs/max98090.c            | 2 +-
 sound/soc/codecs/max98095.c            | 2 +-
 sound/soc/codecs/max98357a.c           | 2 +-
 sound/soc/codecs/max9850.c             | 2 +-
 sound/soc/codecs/max9860.c             | 2 +-
 sound/soc/codecs/max9867.c             | 2 +-
 sound/soc/codecs/max98926.c            | 2 +-
 sound/soc/codecs/mc13783.c             | 2 +-
 sound/soc/codecs/ml26124.c             | 2 +-
 sound/soc/codecs/msm8916-wcd-analog.c  | 2 +-
 sound/soc/codecs/msm8916-wcd-digital.c | 2 +-
 sound/soc/codecs/nau8540.c             | 2 +-
 sound/soc/codecs/nau8810.c             | 2 +-
 sound/soc/codecs/nau8824.c             | 2 +-
 sound/soc/codecs/nau8825.c             | 2 +-
 sound/soc/codecs/pcm1681.c             | 2 +-
 sound/soc/codecs/pcm179x.c             | 2 +-
 sound/soc/codecs/pcm3008.c             | 2 +-
 sound/soc/codecs/pcm512x.c             | 2 +-
 sound/soc/codecs/rt274.c               | 2 +-
 sound/soc/codecs/rt286.c               | 2 +-
 sound/soc/codecs/rt298.c               | 2 +-
 sound/soc/codecs/rt5514.c              | 2 +-
 sound/soc/codecs/rt5616.c              | 2 +-
 sound/soc/codecs/rt5631.c              | 2 +-
 sound/soc/codecs/rt5640.c              | 2 +-
 sound/soc/codecs/rt5645.c              | 2 +-
 sound/soc/codecs/rt5651.c              | 2 +-
 sound/soc/codecs/rt5659.c              | 2 +-
 sound/soc/codecs/rt5660.c              | 2 +-
 sound/soc/codecs/rt5663.c              | 2 +-
 sound/soc/codecs/rt5665.c              | 2 +-
 sound/soc/codecs/rt5670.c              | 2 +-
 sound/soc/codecs/rt5677.c              | 2 +-
 sound/soc/codecs/sgtl5000.c            | 2 +-
 sound/soc/codecs/si476x.c              | 2 +-
 sound/soc/codecs/sirf-audio-codec.c    | 2 +-
 sound/soc/codecs/sn95031.c             | 2 +-
 sound/soc/codecs/spdif_receiver.c      | 2 +-
 sound/soc/codecs/spdif_transmitter.c   | 2 +-
 sound/soc/codecs/ssm2518.c             | 2 +-
 sound/soc/codecs/ssm2602.c             | 2 +-
 sound/soc/codecs/ssm4567.c             | 2 +-
 sound/soc/codecs/stac9766.c            | 2 +-
 sound/soc/codecs/tas2552.c             | 2 +-
 sound/soc/codecs/tas5086.c             | 2 +-
 sound/soc/codecs/tas5720.c             | 2 +-
 sound/soc/codecs/tlv320aic23.c         | 2 +-
 sound/soc/codecs/tlv320aic26.c         | 2 +-
 sound/soc/codecs/tlv320aic31xx.c       | 2 +-
 sound/soc/codecs/tlv320aic32x4.c       | 2 +-
 sound/soc/codecs/tlv320aic3x.c         | 2 +-
 sound/soc/codecs/tlv320dac33.c         | 2 +-
 sound/soc/codecs/twl4030.c             | 2 +-
 sound/soc/codecs/twl6040.c             | 2 +-
 sound/soc/codecs/uda134x.c             | 2 +-
 sound/soc/codecs/uda1380.c             | 2 +-
 sound/soc/codecs/wl1273.c              | 2 +-
 sound/soc/codecs/zx_aud96p22.c         | 2 +-
 109 files changed, 109 insertions(+), 109 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index b013a4c62b63..848c5fe49bc7 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1355,7 +1355,7 @@ static struct regmap *pm860x_get_regmap(struct device *dev)
 	return pm860x->regmap;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
+static const struct snd_soc_codec_driver soc_codec_dev_pm860x = {
 	.probe		= pm860x_probe,
 	.remove		= pm860x_remove,
 	.set_bias_level	= pm860x_set_bias_level,
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 312b2a11abb6..006627b8c3a8 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2523,7 +2523,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 	return status;
 }
 
-static struct snd_soc_codec_driver ab8500_codec_driver = {
+static const struct snd_soc_codec_driver ab8500_codec_driver = {
 	.probe =		ab8500_codec_probe,
 	.component_driver = {
 		.controls =		ab8500_ctrls,
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index f7f04c6be01e..440b4ce54376 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -112,7 +112,7 @@ static int ac97_soc_resume(struct snd_soc_codec *codec)
 #define ac97_soc_resume NULL
 #endif
 
-static struct snd_soc_codec_driver soc_codec_dev_ac97 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ac97 = {
 	.probe = 	ac97_soc_probe,
 	.suspend =	ac97_soc_suspend,
 	.resume =	ac97_soc_resume,
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index a478239aadac..d0361caad09e 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -321,7 +321,7 @@ static int ad1836_remove(struct snd_soc_codec *codec)
 		AD1836_ADC_SERFMT_MASK, 0);
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_ad1836 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ad1836 = {
 	.probe = ad1836_probe,
 	.remove = ad1836_remove,
 	.suspend = ad1836_suspend,
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index d643557d89a7..d10988eec0c1 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -408,7 +408,7 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
+static const struct snd_soc_codec_driver soc_codec_dev_ad193x = {
 	.probe = ad193x_codec_probe,
 	.component_driver = {
 		.controls		= ad193x_snd_controls,
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index b7c1b9f4bf5f..ce89bfb42094 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -295,7 +295,7 @@ static int ad1980_soc_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
 	.probe = 	ad1980_soc_probe,
 	.remove = 	ad1980_soc_remove,
 
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index 3e358a49442d..d8d86a0fea60 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -54,7 +54,7 @@ static struct snd_soc_dai_driver ad73311_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ad73311 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ad73311 = {
 	.component_driver = {
 		.dapm_widgets		= ad73311_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(ad73311_dapm_widgets),
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 8fa9045571ff..a865945d776a 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1458,7 +1458,7 @@ static const struct regmap_config adau1373_regmap_config = {
 	.num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults),
 };
 
-static struct snd_soc_codec_driver adau1373_codec_driver = {
+static const struct snd_soc_codec_driver adau1373_codec_driver = {
 	.probe =	adau1373_probe,
 	.resume =	adau1373_resume,
 	.set_bias_level = adau1373_set_bias_level,
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 3bad1bc8c00a..805afac8146b 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -757,7 +757,7 @@ static int adau1701_resume(struct snd_soc_codec *codec)
 #define adau1701_suspend 	NULL
 #endif /* CONFIG_PM */
 
-static struct snd_soc_codec_driver adau1701_codec_drv = {
+static const struct snd_soc_codec_driver adau1701_codec_drv = {
 	.probe			= adau1701_probe,
 	.remove			= adau1701_remove,
 	.resume			= adau1701_resume,
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index b319db6a69f8..329281675765 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -867,7 +867,7 @@ static int adau1977_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver adau1977_codec_driver = {
+static const struct snd_soc_codec_driver adau1977_codec_driver = {
 	.probe = adau1977_codec_probe,
 	.set_bias_level = adau1977_set_bias_level,
 	.set_sysclk = adau1977_set_sysclk,
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 6e793ebb5883..da7ca81f47cf 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -825,7 +825,7 @@ static int adav80x_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver adav80x_codec_driver = {
+static const struct snd_soc_codec_driver adav80x_codec_driver = {
 	.probe = adav80x_probe,
 	.resume = adav80x_resume,
 	.set_bias_level = adav80x_set_bias_level,
diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c
index 90c756d183b4..b7f0057c0239 100644
--- a/sound/soc/codecs/ads117x.c
+++ b/sound/soc/codecs/ads117x.c
@@ -58,7 +58,7 @@ static struct snd_soc_dai_driver ads117x_dai = {
 		.formats = ADS117X_FORMATS,},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ads117x = {
+static const struct snd_soc_codec_driver soc_codec_dev_ads117x = {
 	.component_driver = {
 		.dapm_widgets		= ads117x_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(ads117x_dapm_widgets),
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index 1a9d233c94d0..dbb184118f2e 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -242,7 +242,7 @@ static int ak4104_soc_resume(struct snd_soc_codec *codec)
 #define ak4104_soc_resume	NULL
 #endif /* CONFIG_PM */
 
-static struct snd_soc_codec_driver soc_codec_device_ak4104 = {
+static const struct snd_soc_codec_driver soc_codec_device_ak4104 = {
 	.probe = ak4104_probe,
 	.remove = ak4104_remove,
 	.suspend = ak4104_soc_suspend,
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 66cfffde9a12..e3c157dc88db 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -390,7 +390,7 @@ static const struct regmap_config ak4535_regmap = {
 	.num_reg_defaults = ARRAY_SIZE(ak4535_reg_defaults),
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ak4535 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ak4535 = {
 	.resume =	ak4535_resume,
 	.set_bias_level = ak4535_set_bias_level,
 	.suspend_bias_off = true,
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
index b92c548b9d29..0bb4fe5c122a 100644
--- a/sound/soc/codecs/ak4554.c
+++ b/sound/soc/codecs/ak4554.c
@@ -64,7 +64,7 @@ static struct snd_soc_dai_driver ak4554_dai = {
 	.symmetric_rates = 1,
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ak4554 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ak4554 = {
 	.component_driver = {
 		.dapm_widgets		= ak4554_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(ak4554_dapm_widgets),
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 690edebf029e..b95bb8b52e51 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -522,7 +522,7 @@ static int ak4613_resume(struct snd_soc_codec *codec)
 	return regcache_sync(regmap);
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
 	.suspend		= ak4613_suspend,
 	.resume			= ak4613_resume,
 	.set_bias_level		= ak4613_set_bias_level,
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index ebdaf56c1d61..60142ff32d4f 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -524,7 +524,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = {
 },
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
 	.component_driver = {
 		.controls		= ak4641_snd_controls,
 		.num_controls		= ARRAY_SIZE(ak4641_snd_controls),
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 66de8a2013a6..29530c567bd9 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -550,7 +550,7 @@ static int ak4642_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
 	.probe			= ak4642_probe,
 	.suspend		= ak4642_suspend,
 	.resume			= ak4642_resume,
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 6088afaabf62..dcfdff56fc5a 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -610,7 +610,7 @@ static struct snd_soc_dai_driver ak4671_dai = {
 	.ops = &ak4671_dai_ops,
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ak4671 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ak4671 = {
 	.set_bias_level = ak4671_set_bias_level,
 	.component_driver = {
 		.controls		= ak4671_snd_controls,
diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c
index 0ef2df223336..d0e16c03815c 100644
--- a/sound/soc/codecs/ak5386.c
+++ b/sound/soc/codecs/ak5386.c
@@ -69,7 +69,7 @@ static int ak5386_soc_resume(struct snd_soc_codec *codec)
 #define ak5386_soc_resume	NULL
 #endif /* CONFIG_PM */
 
-static struct snd_soc_codec_driver soc_codec_ak5386 = {
+static const struct snd_soc_codec_driver soc_codec_ak5386 = {
 	.probe = ak5386_soc_probe,
 	.remove = ak5386_soc_remove,
 	.suspend = ak5386_soc_suspend,
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index d2e3a3ef7499..1db965a93632 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -951,7 +951,7 @@ static int alc5623_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
+static const struct snd_soc_codec_driver soc_codec_device_alc5623 = {
 	.probe = alc5623_probe,
 	.suspend = alc5623_suspend,
 	.resume = alc5623_resume,
diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c
index 8014e697d540..806191addb44 100644
--- a/sound/soc/codecs/bt-sco.c
+++ b/sound/soc/codecs/bt-sco.c
@@ -62,7 +62,7 @@ static struct snd_soc_dai_driver bt_sco_dai[] = {
 	}
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_bt_sco = {
+static const struct snd_soc_codec_driver soc_codec_dev_bt_sco = {
 	.component_driver = {
 		.dapm_widgets		= bt_sco_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(bt_sco_widgets),
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 7a2d9a2ee427..6ed2cc374768 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -128,7 +128,7 @@ static struct regmap *cq93vc_get_regmap(struct device *dev)
 	return davinci_vc->regmap;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_cq93vc = {
+static const struct snd_soc_codec_driver soc_codec_dev_cq93vc = {
 	.set_bias_level = cq93vc_set_bias_level,
 	.get_regmap = cq93vc_get_regmap,
 	.component_driver = {
diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c
index 6df29fa30fb9..61d128b53e14 100644
--- a/sound/soc/codecs/cs35l33.c
+++ b/sound/soc/codecs/cs35l33.c
@@ -831,7 +831,7 @@ static int cs35l33_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs35l33 = {
 	.probe = cs35l33_probe,
 
 	.set_bias_level = cs35l33_set_bias_level,
diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c
index 0a747c66cc6c..15d712f4803e 100644
--- a/sound/soc/codecs/cs35l34.c
+++ b/sound/soc/codecs/cs35l34.c
@@ -779,7 +779,7 @@ static int cs35l34_probe(struct snd_soc_codec *codec)
 }
 
 
-static struct snd_soc_codec_driver soc_codec_dev_cs35l34 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs35l34 = {
 	.probe = cs35l34_probe,
 
 	.component_driver = {
diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c
index f1ee184ecab2..129978d1243e 100644
--- a/sound/soc/codecs/cs35l35.c
+++ b/sound/soc/codecs/cs35l35.c
@@ -1079,7 +1079,7 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_cs35l35 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs35l35 = {
 	.probe = cs35l35_codec_probe,
 	.set_sysclk = cs35l35_codec_set_sysclk,
 	.component_driver = {
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index d8824773dc29..49a80627af12 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -639,7 +639,7 @@ static int cs4271_codec_remove(struct snd_soc_codec *codec)
 	return 0;
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
 	.probe			= cs4271_codec_probe,
 	.remove			= cs4271_codec_remove,
 	.suspend		= cs4271_soc_suspend,
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 96cfe38943cd..f8072f1897d4 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -504,7 +504,7 @@ static int cs42l51_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
+static const struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
 	.probe = cs42l51_codec_probe,
 
 	.component_driver = {
diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index 231ca935cdf3..0a749c79ef57 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -255,7 +255,7 @@ static struct snd_soc_dai_driver cs4349_dai = {
 	.symmetric_rates = 1,
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_cs4349 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs4349 = {
 	.component_driver = {
 		.controls		= cs4349_snd_controls,
 		.num_controls		= ARRAY_SIZE(cs4349_snd_controls),
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 47e6fddef92b..d323caa9c816 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -1183,7 +1183,7 @@ static struct regmap *cs47l24_get_regmap(struct device *dev)
 	return priv->core.arizona->regmap;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_cs47l24 = {
+static const struct snd_soc_codec_driver soc_codec_dev_cs47l24 = {
 	.probe = cs47l24_codec_probe,
 	.remove = cs47l24_codec_remove,
 	.get_regmap = cs47l24_get_regmap,
diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c
index 06933a5d0a75..da4ee5633778 100644
--- a/sound/soc/codecs/cs53l30.c
+++ b/sound/soc/codecs/cs53l30.c
@@ -892,7 +892,7 @@ static int cs53l30_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver cs53l30_driver = {
+static const struct snd_soc_codec_driver cs53l30_driver = {
 	.probe = cs53l30_codec_probe,
 	.set_bias_level = cs53l30_set_bias_level,
 	.idle_bias_off = true,
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 2c12471e42a6..46b1fbb66eba 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -398,7 +398,7 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
 
 static const u8 cx20442_reg;
 
-static struct snd_soc_codec_driver cx20442_codec_dev = {
+static const struct snd_soc_codec_driver cx20442_codec_dev = {
 	.probe = 	cx20442_codec_probe,
 	.remove = 	cx20442_codec_remove,
 	.set_bias_level = cx20442_set_bias_level,
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 17053dfc94cf..1af443ccbc51 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1164,7 +1164,7 @@ static int da7210_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
+static const struct snd_soc_codec_driver soc_codec_dev_da7210 = {
 	.probe			= da7210_probe,
 
 	.component_driver = {
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index c3e11897f8ae..cc0b2d2eaf15 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1787,7 +1787,7 @@ static int da7213_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_da7213 = {
+static const struct snd_soc_codec_driver soc_codec_dev_da7213 = {
 	.probe			= da7213_probe,
 	.set_bias_level		= da7213_set_bias_level,
 
diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c
index 6e1940eb0653..b2d42ec1dcd9 100644
--- a/sound/soc/codecs/da7218.c
+++ b/sound/soc/codecs/da7218.c
@@ -3035,7 +3035,7 @@ static int da7218_resume(struct snd_soc_codec *codec)
 #define da7218_resume NULL
 #endif
 
-static struct snd_soc_codec_driver soc_codec_dev_da7218 = {
+static const struct snd_soc_codec_driver soc_codec_dev_da7218 = {
 	.probe			= da7218_probe,
 	.remove			= da7218_remove,
 	.suspend		= da7218_suspend,
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index f71d72c22bfc..6f088536df32 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -1891,7 +1891,7 @@ static int da7219_resume(struct snd_soc_codec *codec)
 #define da7219_resume NULL
 #endif
 
-static struct snd_soc_codec_driver soc_codec_dev_da7219 = {
+static const struct snd_soc_codec_driver soc_codec_dev_da7219 = {
 	.probe			= da7219_probe,
 	.remove			= da7219_remove,
 	.suspend		= da7219_suspend,
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index c1cc1c1c28f2..83db4d23c90b 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1499,7 +1499,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_da732x = {
+static const struct snd_soc_codec_driver soc_codec_dev_da732x = {
 	.set_bias_level		= da732x_set_bias_level,
 	.component_driver = {
 		.controls		= da732x_snd_controls,
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 4efb5f897a0c..bd7faaee5802 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1451,7 +1451,7 @@ static int da9055_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_da9055 = {
+static const struct snd_soc_codec_driver soc_codec_dev_da9055 = {
 	.probe			= da9055_probe,
 	.set_bias_level		= da9055_set_bias_level,
 
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index c82b9dc41e9a..6fe6c0ac4f0b 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,7 +50,7 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"DMIC AIF", NULL, "DMic"},
 };
 
-static struct snd_soc_codec_driver soc_dmic = {
+static const struct snd_soc_codec_driver soc_dmic = {
 	.component_driver = {
 		.dapm_widgets		= dmic_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(dmic_dapm_widgets),
diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c
index 25ede825d349..3869025754d8 100644
--- a/sound/soc/codecs/es7134.c
+++ b/sound/soc/codecs/es7134.c
@@ -69,7 +69,7 @@ static const struct snd_soc_dapm_route es7134_dapm_routes[] = {
 	{ "AOUTR", NULL, "DAC" },
 };
 
-static struct snd_soc_codec_driver es7134_codec_driver = {
+static const struct snd_soc_codec_driver es7134_codec_driver = {
 	.component_driver = {
 		.dapm_widgets		= es7134_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(es7134_dapm_widgets),
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index ecc02449c569..4f35af6a5d3a 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -554,7 +554,7 @@ static int es8316_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_es8316 = {
+static const struct snd_soc_codec_driver soc_codec_dev_es8316 = {
 	.probe		= es8316_probe,
 	.idle_bias_off	= true,
 
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index ed7cc42d1ee2..bcdb8914ec16 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -830,7 +830,7 @@ const struct regmap_config es8328_regmap_config = {
 };
 EXPORT_SYMBOL_GPL(es8328_regmap_config);
 
-static struct snd_soc_codec_driver es8328_codec_driver = {
+static const struct snd_soc_codec_driver es8328_codec_driver = {
 	.probe		  = es8328_codec_probe,
 	.suspend	  = es8328_suspend,
 	.resume		  = es8328_resume,
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index e6de50acefd4..e808f94bf8d1 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1858,7 +1858,7 @@ static void hdmi_codec_complete(struct device *dev)
 #define hdmi_codec_complete NULL
 #endif
 
-static struct snd_soc_codec_driver hdmi_hda_codec = {
+static const struct snd_soc_codec_driver hdmi_hda_codec = {
 	.probe		= hdmi_codec_probe,
 	.remove		= hdmi_codec_remove,
 	.idle_bias_off	= true,
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 509ab513b4b2..f288404f0a61 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -733,7 +733,7 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component,
 	return ret;
 }
 
-static struct snd_soc_codec_driver hdmi_codec = {
+static const struct snd_soc_codec_driver hdmi_codec = {
 	.component_driver = {
 		.dapm_widgets		= hdmi_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(hdmi_widgets),
diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c
index dd850b93938d..651206273f36 100644
--- a/sound/soc/codecs/ics43432.c
+++ b/sound/soc/codecs/ics43432.c
@@ -37,7 +37,7 @@ static struct snd_soc_dai_driver ics43432_dai = {
 	},
 };
 
-static struct snd_soc_codec_driver ics43432_codec_driver = {
+static const struct snd_soc_codec_driver ics43432_codec_driver = {
 };
 
 static int ics43432_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c
index b918ba5c8ce5..64b0be94bea3 100644
--- a/sound/soc/codecs/inno_rk3036.c
+++ b/sound/soc/codecs/inno_rk3036.c
@@ -376,7 +376,7 @@ static int rk3036_codec_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static struct snd_soc_codec_driver rk3036_codec_driver = {
+static const struct snd_soc_codec_driver rk3036_codec_driver = {
 	.probe			= rk3036_codec_probe,
 	.remove			= rk3036_codec_remove,
 	.set_bias_level		= rk3036_codec_set_bias_level,
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index a4b0eded984a..5ca99280ae00 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -1087,7 +1087,7 @@ static struct snd_soc_dai_driver isabelle_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_isabelle = {
+static const struct snd_soc_codec_driver soc_codec_dev_isabelle = {
 	.set_bias_level = isabelle_set_bias_level,
 	.component_driver = {
 		.controls		= isabelle_snd_controls,
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 0290fab383da..6324ccdc8a5c 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -293,7 +293,7 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = {
+static const struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = {
 	.probe = jz4740_codec_dev_probe,
 	.set_bias_level = jz4740_codec_set_bias_level,
 	.suspend_bias_off = true,
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 8d413c2677cc..41e09d1287b8 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1389,7 +1389,7 @@ static struct snd_soc_dai_driver lm49453_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
+static const struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
 	.set_bias_level = lm49453_set_bias_level,
 	.component_driver = {
 		.controls		= lm49453_snd_controls,
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 72f77455582e..f0bb830874e5 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1698,7 +1698,7 @@ static int max98088_remove(struct snd_soc_codec *codec)
        return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_max98088 = {
+static const struct snd_soc_codec_driver soc_codec_dev_max98088 = {
 	.probe   = max98088_probe,
 	.remove  = max98088_remove,
 	.set_bias_level = max98088_set_bias_level,
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 66828480d484..13bcfb1ef9b4 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2499,7 +2499,7 @@ static void max98090_seq_notifier(struct snd_soc_dapm_context *dapm,
 	}
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_max98090 = {
+static const struct snd_soc_codec_driver soc_codec_dev_max98090 = {
 	.probe   = max98090_probe,
 	.remove  = max98090_remove,
 	.seq_notifier = max98090_seq_notifier,
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 6f8a757876ed..5ead87d2ab1d 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -2102,7 +2102,7 @@ static int max98095_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_max98095 = {
+static const struct snd_soc_codec_driver soc_codec_dev_max98095 = {
 	.probe   = max98095_probe,
 	.remove  = max98095_remove,
 	.suspend = max98095_suspend,
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 6a6b68a4cb52..426ed2dae6ca 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -72,7 +72,7 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver max98357a_codec_driver = {
+static const struct snd_soc_codec_driver max98357a_codec_driver = {
 	.probe			= max98357a_codec_probe,
 	.component_driver = {
 		.dapm_widgets		= max98357a_dapm_widgets,
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 0610840733d1..a3dfc918c278 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -301,7 +301,7 @@ static int max9850_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_max9850 = {
+static const struct snd_soc_codec_driver soc_codec_dev_max9850 = {
 	.probe =	max9850_probe,
 	.set_bias_level = max9850_set_bias_level,
 	.suspend_bias_off = true,
diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c
index 499bdbfd0a2d..a2dc6a47f466 100644
--- a/sound/soc/codecs/max9860.c
+++ b/sound/soc/codecs/max9860.c
@@ -534,7 +534,7 @@ static int max9860_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static struct snd_soc_codec_driver max9860_codec_driver = {
+static const struct snd_soc_codec_driver max9860_codec_driver = {
 	.set_bias_level = max9860_set_bias_level,
 	.idle_bias_off = true,
 
diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index 2a40a69a7513..6c0c0d6e8f3c 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -413,7 +413,7 @@ static int max9867_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver max9867_codec = {
+static const struct snd_soc_codec_driver max9867_codec = {
 	.probe = max9867_probe,
 	.component_driver = {
 		.controls		= max9867_snd_controls,
diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
index 1eff7e0b092e..7a39bfb9e0f9 100644
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -496,7 +496,7 @@ static int max98926_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_max98926 = {
+static const struct snd_soc_codec_driver soc_codec_dev_max98926 = {
 	.probe	= max98926_probe,
 	.component_driver = {
 		.controls		= max98926_snd_controls,
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 90562703dcfd..4fd8d1dc4eef 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -733,7 +733,7 @@ static struct regmap *mc13783_get_regmap(struct device *dev)
 	return dev_get_regmap(dev->parent, NULL);
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
+static const struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
 	.probe		= mc13783_probe,
 	.remove		= mc13783_remove,
 	.get_regmap	= mc13783_get_regmap,
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 69e5e18880c5..5cc960d8211e 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -537,7 +537,7 @@ static int ml26124_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
 	.probe =	ml26124_probe,
 	.set_bias_level = ml26124_set_bias_level,
 	.suspend_bias_off = true,
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index a78802920c3c..4cc52cecf3b7 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -785,7 +785,7 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = {
 	       },
 };
 
-static struct snd_soc_codec_driver pm8916_wcd_analog = {
+static const struct snd_soc_codec_driver pm8916_wcd_analog = {
 	.probe = pm8916_wcd_analog_probe,
 	.remove = pm8916_wcd_analog_remove,
 	.get_regmap = pm8916_get_regmap,
diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 661cd6dd5473..43a915c725a0 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -830,7 +830,7 @@ static struct snd_soc_dai_driver msm8916_wcd_digital_dai[] = {
 	       },
 };
 
-static struct snd_soc_codec_driver msm8916_wcd_digital = {
+static const struct snd_soc_codec_driver msm8916_wcd_digital = {
 	.probe = msm8916_wcd_digital_codec_probe,
 	.set_sysclk = msm8916_wcd_digital_codec_set_sysclk,
 	.component_driver = {
diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c
index c8bcb1db966d..f9c9933acffb 100644
--- a/sound/soc/codecs/nau8540.c
+++ b/sound/soc/codecs/nau8540.c
@@ -735,7 +735,7 @@ static int __maybe_unused nau8540_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver nau8540_codec_driver = {
+static const struct snd_soc_codec_driver nau8540_codec_driver = {
 	.set_sysclk = nau8540_set_sysclk,
 	.set_pll = nau8540_set_pll,
 	.suspend = nau8540_suspend,
diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c
index e45518629968..c8e2451ae0a3 100644
--- a/sound/soc/codecs/nau8810.c
+++ b/sound/soc/codecs/nau8810.c
@@ -808,7 +808,7 @@ static const struct regmap_config nau8810_regmap_config = {
 	.num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults),
 };
 
-static struct snd_soc_codec_driver nau8810_codec_driver = {
+static const struct snd_soc_codec_driver nau8810_codec_driver = {
 	.set_bias_level = nau8810_set_bias_level,
 	.suspend_bias_off = true,
 
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 3a309b18035e..0240759f951c 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -1469,7 +1469,7 @@ static int __maybe_unused nau8824_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver nau8824_codec_driver = {
+static const struct snd_soc_codec_driver nau8824_codec_driver = {
 	.probe = nau8824_codec_probe,
 	.set_sysclk = nau8824_set_sysclk,
 	.set_pll = nau8824_set_pll,
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 46a30eaa7ace..000aa79314fa 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -2388,7 +2388,7 @@ static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver nau8825_codec_driver = {
+static const struct snd_soc_codec_driver nau8825_codec_driver = {
 	.probe = nau8825_codec_probe,
 	.remove = nau8825_codec_remove,
 	.set_sysclk = nau8825_set_sysclk,
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 0b14efab6280..c7e28dd2e815 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -288,7 +288,7 @@ static const struct regmap_config pcm1681_regmap = {
 	.readable_reg		= pcm1681_accessible_reg,
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = {
+static const struct snd_soc_codec_driver soc_codec_dev_pcm1681 = {
 	.component_driver = {
 		.controls		= pcm1681_controls,
 		.num_controls		= ARRAY_SIZE(pcm1681_controls),
diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c
index b813a154ddd9..82a3d9db32cb 100644
--- a/sound/soc/codecs/pcm179x.c
+++ b/sound/soc/codecs/pcm179x.c
@@ -205,7 +205,7 @@ const struct regmap_config pcm179x_regmap_config = {
 };
 EXPORT_SYMBOL_GPL(pcm179x_regmap_config);
 
-static struct snd_soc_codec_driver soc_codec_dev_pcm179x = {
+static const struct snd_soc_codec_driver soc_codec_dev_pcm179x = {
 	.component_driver = {
 		.controls		= pcm179x_controls,
 		.num_controls		= ARRAY_SIZE(pcm179x_controls),
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index 708af05486f6..e59d8ffb93bd 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -98,7 +98,7 @@ static struct snd_soc_dai_driver pcm3008_dai = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = {
+static const struct snd_soc_codec_driver soc_codec_dev_pcm3008 = {
 	.component_driver = {
 		.dapm_widgets		= pcm3008_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(pcm3008_dapm_widgets),
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 72b19e62f626..f1005a31c709 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -1344,7 +1344,7 @@ static struct snd_soc_dai_driver pcm512x_dai = {
 	.ops = &pcm512x_dai_ops,
 };
 
-static struct snd_soc_codec_driver pcm512x_codec_driver = {
+static const struct snd_soc_codec_driver pcm512x_codec_driver = {
 	.set_bias_level = pcm512x_set_bias_level,
 	.idle_bias_off = true,
 
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index fb683ffc4e03..58cfe244f81e 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -1055,7 +1055,7 @@ static struct snd_soc_dai_driver rt274_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt274 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt274 = {
 	.probe = rt274_probe,
 	.remove = rt274_remove,
 	.suspend = rt274_suspend,
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 7899a2cdeb42..af6325c78292 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -1046,7 +1046,7 @@ static struct snd_soc_dai_driver rt286_dai[] = {
 
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt286 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt286 = {
 	.probe = rt286_probe,
 	.remove = rt286_remove,
 	.suspend = rt286_suspend,
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index d9e96e65e1c4..ce963768449f 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -1113,7 +1113,7 @@ static struct snd_soc_dai_driver rt298_dai[] = {
 
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt298 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt298 = {
 	.probe = rt298_probe,
 	.remove = rt298_remove,
 	.suspend = rt298_suspend,
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 1b6796c4c471..7a1b36f6596a 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -1042,7 +1042,7 @@ struct snd_soc_dai_driver rt5514_dai[] = {
 	}
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5514 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5514 = {
 	.probe = rt5514_probe,
 	.idle_bias_off = true,
 	.set_bias_level = rt5514_set_bias_level,
diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c
index 7d6e0823f98f..3c5f555ae6d0 100644
--- a/sound/soc/codecs/rt5616.c
+++ b/sound/soc/codecs/rt5616.c
@@ -1294,7 +1294,7 @@ static struct snd_soc_dai_driver rt5616_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5616 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5616 = {
 	.probe = rt5616_probe,
 	.suspend = rt5616_suspend,
 	.resume = rt5616_resume,
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 0e418089c053..55b04c55fb4b 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1653,7 +1653,7 @@ static struct snd_soc_dai_driver rt5631_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5631 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5631 = {
 	.probe = rt5631_probe,
 	.set_bias_level = rt5631_set_bias_level,
 	.suspend_bias_off = true,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 1584ccc3a87b..438fe52a12df 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2259,7 +2259,7 @@ static struct snd_soc_dai_driver rt5640_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
 	.probe = rt5640_probe,
 	.remove = rt5640_remove,
 	.suspend = rt5640_suspend,
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 9ec58166f7c4..ce31d0dcf894 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3473,7 +3473,7 @@ static struct snd_soc_dai_driver rt5645_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5645 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5645 = {
 	.probe = rt5645_probe,
 	.remove = rt5645_remove,
 	.suspend = rt5645_suspend,
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index db05b60d5002..da60b28ba3df 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1664,7 +1664,7 @@ static struct snd_soc_dai_driver rt5651_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5651 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5651 = {
 	.probe = rt5651_probe,
 	.suspend = rt5651_suspend,
 	.resume = rt5651_resume,
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 1b7060850340..e8018fce7e4b 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3730,7 +3730,7 @@ static struct snd_soc_dai_driver rt5659_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5659 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5659 = {
 	.probe = rt5659_probe,
 	.remove = rt5659_remove,
 	.suspend = rt5659_suspend,
diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c
index c93490d77f2a..d22ef00e0d96 100644
--- a/sound/soc/codecs/rt5660.c
+++ b/sound/soc/codecs/rt5660.c
@@ -1197,7 +1197,7 @@ static struct snd_soc_dai_driver rt5660_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5660 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5660 = {
 	.probe = rt5660_probe,
 	.remove = rt5660_remove,
 	.suspend = rt5660_suspend,
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index a33202affeb1..ebc5e3b2b62b 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -2891,7 +2891,7 @@ static struct snd_soc_dai_driver rt5663_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5663 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5663 = {
 	.probe = rt5663_probe,
 	.remove = rt5663_remove,
 	.suspend = rt5663_suspend,
diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index 370ed54d1e15..2c405456d89f 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -4562,7 +4562,7 @@ static struct snd_soc_dai_driver rt5665_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5665 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5665 = {
 	.probe = rt5665_probe,
 	.remove = rt5665_remove,
 	.suspend = rt5665_suspend,
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 0ec7985ed306..52a6ce8f0b39 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2765,7 +2765,7 @@ static struct snd_soc_dai_driver rt5670_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5670 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5670 = {
 	.probe = rt5670_probe,
 	.remove = rt5670_remove,
 	.suspend = rt5670_suspend,
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 36e530a36c82..ce9b65d466e8 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4968,7 +4968,7 @@ static struct snd_soc_dai_driver rt5677_dai[] = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_rt5677 = {
+static const struct snd_soc_codec_driver soc_codec_dev_rt5677 = {
 	.probe = rt5677_probe,
 	.remove = rt5677_remove,
 	.suspend = rt5677_suspend,
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 8f6814c1eb6b..68e7020b2875 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1248,7 +1248,7 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver sgtl5000_driver = {
+static const struct snd_soc_codec_driver sgtl5000_driver = {
 	.probe = sgtl5000_probe,
 	.remove = sgtl5000_remove,
 	.set_bias_level = sgtl5000_set_bias_level,
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index 5344f4aa8fde..354dc0d64f11 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -236,7 +236,7 @@ static struct regmap *si476x_get_regmap(struct device *dev)
 	return dev_get_regmap(dev->parent, NULL);
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_si476x = {
+static const struct snd_soc_codec_driver soc_codec_dev_si476x = {
 	.get_regmap = si476x_get_regmap,
 	.component_driver = {
 		.dapm_widgets		= si476x_dapm_widgets,
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
index 6bfd25c289d1..50bc22266ecb 100644
--- a/sound/soc/codecs/sirf-audio-codec.c
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -429,7 +429,7 @@ static int sirf_audio_codec_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
+static const struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
 	.probe = sirf_audio_codec_probe,
 	.remove = sirf_audio_codec_remove,
 	.dapm_widgets = sirf_audio_codec_dapm_widgets,
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index eae54c37cff9..887923e68849 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -883,7 +883,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver sn95031_codec = {
+static const struct snd_soc_codec_driver sn95031_codec = {
 	.probe		= sn95031_codec_probe,
 	.set_bias_level	= sn95031_set_vaud_bias,
 	.idle_bias_off	= true,
diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c
index 234f87b54838..7acd05140a81 100644
--- a/sound/soc/codecs/spdif_receiver.c
+++ b/sound/soc/codecs/spdif_receiver.c
@@ -37,7 +37,7 @@ static const struct snd_soc_dapm_route dir_routes[] = {
 			SNDRV_PCM_FMTBIT_S24_LE | \
 			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
 
-static struct snd_soc_codec_driver soc_codec_spdif_dir = {
+static const struct snd_soc_codec_driver soc_codec_spdif_dir = {
 	.component_driver = {
 		.dapm_widgets		= dir_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(dir_widgets),
diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c
index ee367536a498..063a64ff82d3 100644
--- a/sound/soc/codecs/spdif_transmitter.c
+++ b/sound/soc/codecs/spdif_transmitter.c
@@ -37,7 +37,7 @@ static const struct snd_soc_dapm_route dit_routes[] = {
 	{ "spdif-out", NULL, "Playback" },
 };
 
-static struct snd_soc_codec_driver soc_codec_spdif_dit = {
+static const struct snd_soc_codec_driver soc_codec_spdif_dit = {
 	.component_driver = {
 		.dapm_widgets		= dit_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(dit_widgets),
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index 38a85f3adc80..15486fd16269 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -710,7 +710,7 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			SSM2518_POWER1_NO_BCLK, val);
 }
 
-static struct snd_soc_codec_driver ssm2518_codec_driver = {
+static const struct snd_soc_codec_driver ssm2518_codec_driver = {
 	.set_bias_level = ssm2518_set_bias_level,
 	.set_sysclk = ssm2518_set_sysclk,
 	.idle_bias_off = true,
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 993bde29ca1b..9b341c23f62b 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -591,7 +591,7 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec)
 	return ret;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
+static const struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
 	.probe =	ssm260x_codec_probe,
 	.resume =	ssm2602_resume,
 	.set_bias_level = ssm2602_set_bias_level,
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index a622623e8558..4afeddef7728 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -417,7 +417,7 @@ static struct snd_soc_dai_driver ssm4567_dai = {
 	.ops = &ssm4567_dai_ops,
 };
 
-static struct snd_soc_codec_driver ssm4567_codec_driver = {
+static const struct snd_soc_codec_driver ssm4567_codec_driver = {
 	.set_bias_level = ssm4567_set_bias_level,
 	.idle_bias_off = true,
 
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 9de7fe8af255..c66363a2cac7 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -307,7 +307,7 @@ static int stac9766_codec_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
+static const struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
 	.component_driver = {
 		.controls		= stac9766_snd_ac97_controls,
 		.num_controls		= ARRAY_SIZE(stac9766_snd_ac97_controls),
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 8840f72f3c4a..49cf9bc32eb6 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -660,7 +660,7 @@ static int tas2552_resume(struct snd_soc_codec *codec)
 #define tas2552_resume NULL
 #endif
 
-static struct snd_soc_codec_driver soc_codec_dev_tas2552 = {
+static const struct snd_soc_codec_driver soc_codec_dev_tas2552 = {
 	.probe = tas2552_codec_probe,
 	.remove = tas2552_codec_remove,
 	.suspend =	tas2552_suspend,
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
index b7de857abb16..199272d5cb6a 100644
--- a/sound/soc/codecs/tas5086.c
+++ b/sound/soc/codecs/tas5086.c
@@ -885,7 +885,7 @@ static int tas5086_remove(struct snd_soc_codec *codec)
 	return 0;
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_tas5086 = {
+static const struct snd_soc_codec_driver soc_codec_dev_tas5086 = {
 	.probe			= tas5086_probe,
 	.remove			= tas5086_remove,
 	.suspend		= tas5086_soc_suspend,
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index c65b917598d2..d14b78b1917b 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -483,7 +483,7 @@ static const struct snd_soc_dapm_route tas5720_audio_map[] = {
 	{ "OUT", NULL, "DAC" },
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
+static const struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
 	.probe = tas5720_codec_probe,
 	.remove = tas5720_codec_remove,
 	.suspend = tas5720_suspend,
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 628a8eeaab68..3d42138a7974 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -576,7 +576,7 @@ static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
+static const struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
 	.probe = tlv320aic23_codec_probe,
 	.resume = tlv320aic23_resume,
 	.set_bias_level = tlv320aic23_set_bias_level,
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 14aa96d41719..89421caaeb70 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -319,7 +319,7 @@ static int aic26_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver aic26_soc_codec_dev = {
+static const struct snd_soc_codec_driver aic26_soc_codec_dev = {
 	.probe = aic26_probe,
 	.component_driver = {
 		.controls		= aic26_snd_controls,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index d7d03c92cb8a..54a87a905eb6 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1185,7 +1185,7 @@ static int aic31xx_codec_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
+static const struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
 	.probe			= aic31xx_codec_probe,
 	.remove			= aic31xx_codec_remove,
 	.set_bias_level		= aic31xx_set_bias_level,
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 28fdfc5ec544..ccfc955321ae 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -792,7 +792,7 @@ static int aic32x4_codec_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
+static const struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 	.probe = aic32x4_codec_probe,
 	.set_bias_level = aic32x4_set_bias_level,
 	.suspend_bias_off = true,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 29bf8c81ae02..405f4602888a 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1684,7 +1684,7 @@ static int aic3x_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
+static const struct snd_soc_codec_driver soc_codec_dev_aic3x = {
 	.set_bias_level = aic3x_set_bias_level,
 	.idle_bias_off = true,
 	.probe = aic3x_probe,
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 7bcf01efdf9a..5b94a151539c 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1433,7 +1433,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = {
+static const struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = {
 	.read = dac33_read_reg_cache,
 	.write = dac33_write_locked,
 	.set_bias_level = dac33_set_bias_level,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index a2104d68169d..d439c4c6fe50 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -2191,7 +2191,7 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
+static const struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
 	.probe = twl4030_soc_probe,
 	.remove = twl4030_soc_remove,
 	.read = twl4030_read,
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 2b6ad09e0886..0dc21f7e0af9 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1155,7 +1155,7 @@ static int twl6040_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_twl6040 = {
+static const struct snd_soc_codec_driver soc_codec_dev_twl6040 = {
 	.probe = twl6040_probe,
 	.remove = twl6040_remove,
 	.read = twl6040_read,
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 5fdee874406d..77c9cc4467b8 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -518,7 +518,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
+static const struct snd_soc_codec_driver soc_codec_dev_uda134x = {
 	.probe =        uda134x_soc_probe,
 	.set_bias_level = uda134x_set_bias_level,
 	.suspend_bias_off = true,
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 61cdc79840e7..926c81ae8185 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -720,7 +720,7 @@ static int uda1380_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_uda1380 = {
+static const struct snd_soc_codec_driver soc_codec_dev_uda1380 = {
 	.probe =	uda1380_probe,
 	.read =		uda1380_read_reg_cache,
 	.write =	uda1380_write,
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index fcffb6e707d9..942f1644973e 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -480,7 +480,7 @@ static int wl1273_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
+static const struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
 	.probe = wl1273_probe,
 	.remove = wl1273_remove,
 
diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c
index 032fb7cf6cbd..36d0a7087a54 100644
--- a/sound/soc/codecs/zx_aud96p22.c
+++ b/sound/soc/codecs/zx_aud96p22.c
@@ -261,7 +261,7 @@ static const struct snd_soc_dapm_route aud96p22_dapm_routes[] = {
 	{ "LINEOUTMN", NULL, "LD2" },
 };
 
-static struct snd_soc_codec_driver aud96p22_driver = {
+static const struct snd_soc_codec_driver aud96p22_driver = {
 	.component_driver = {
 		.controls = aud96p22_snd_controls,
 		.num_controls = ARRAY_SIZE(aud96p22_snd_controls),
-- 
cgit v1.2.3


From 7a0cf42edd9cc335572aa8614df0ebefa2f537be Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Thu, 10 Aug 2017 16:21:34 +0200
Subject: ASoC: compress: Delete error messages for a failed memory allocation
 in snd_soc_new_compress()

Omit extra messages for a memory allocation failure in this function.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-compress.c | 7 ++-----
 1 file changed, 2 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 206f36bf43e8..6c867d50e036 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -764,15 +764,12 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 		direction = SND_COMPRESS_CAPTURE;
 
 	compr = kzalloc(sizeof(*compr), GFP_KERNEL);
-	if (compr == NULL) {
-		snd_printk(KERN_ERR "Cannot allocate compr\n");
+	if (!compr)
 		return -ENOMEM;
-	}
 
 	compr->ops = devm_kzalloc(rtd->card->dev, sizeof(soc_compr_ops),
 				  GFP_KERNEL);
-	if (compr->ops == NULL) {
-		dev_err(rtd->card->dev, "Cannot allocate compressed ops\n");
+	if (!compr->ops) {
 		ret = -ENOMEM;
 		goto compr_err;
 	}
-- 
cgit v1.2.3


From 52981e29eedf6e90ee381c2c1a64be1848d3353e Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Wed, 9 Aug 2017 18:49:23 +0200
Subject: ASoC: codecs: msm8916-wcd-analog: move codec reset to probe

This patch move the codec reset code from dai ops to codec probe, so
that the codec is not held in reset when headset detection block is
still active.

Without this patch the codec block will be in reset as long as its not
actively used, which means headset events will not be functional if the
codec dai is not actively used. Point to note is that the headset
detection blocks will work in low power when there is no active audio
usecase and switch to micbias source when audio usecase is active.

Existing dapms should put the codec in low power state anyway when there
is no audio usecase.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-analog.c | 30 ++++++------------------------
 1 file changed, 6 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index a78802920c3c..f9a74ff2ce99 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -536,6 +536,9 @@ static int pm8916_wcd_analog_probe(struct snd_soc_codec *codec)
 		snd_soc_write(codec, wcd_reg_defaults_2_0[reg].reg,
 			      wcd_reg_defaults_2_0[reg].def);
 
+	snd_soc_update_bits(codec, CDC_D_CDC_RST_CTL,
+			    RST_CTL_DIG_SW_RST_N_MASK,
+			    RST_CTL_DIG_SW_RST_N_REMOVE_RESET);
 	return 0;
 }
 
@@ -543,6 +546,9 @@ static int pm8916_wcd_analog_remove(struct snd_soc_codec *codec)
 {
 	struct pm8916_wcd_analog_priv *priv = dev_get_drvdata(codec->dev);
 
+	snd_soc_update_bits(codec, CDC_D_CDC_RST_CTL,
+			    RST_CTL_DIG_SW_RST_N_MASK, 0);
+
 	return regulator_bulk_disable(ARRAY_SIZE(priv->supplies),
 				      priv->supplies);
 }
@@ -736,28 +742,6 @@ static struct regmap *pm8916_get_regmap(struct device *dev)
 	return dev_get_regmap(dev->parent, NULL);
 }
 
-static int pm8916_wcd_analog_startup(struct snd_pcm_substream *substream,
-				      struct snd_soc_dai *dai)
-{
-	snd_soc_update_bits(dai->codec, CDC_D_CDC_RST_CTL,
-			    RST_CTL_DIG_SW_RST_N_MASK,
-			    RST_CTL_DIG_SW_RST_N_REMOVE_RESET);
-
-	return 0;
-}
-
-static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream,
-					 struct snd_soc_dai *dai)
-{
-	snd_soc_update_bits(dai->codec, CDC_D_CDC_RST_CTL,
-			    RST_CTL_DIG_SW_RST_N_MASK, 0);
-}
-
-static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = {
-	.startup = pm8916_wcd_analog_startup,
-	.shutdown = pm8916_wcd_analog_shutdown,
-};
-
 static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = {
 	[0] = {
 	       .name = "pm8916_wcd_analog_pdm_rx",
@@ -769,7 +753,6 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = {
 			    .channels_min = 1,
 			    .channels_max = 3,
 			    },
-	       .ops = &pm8916_wcd_analog_dai_ops,
 	       },
 	[1] = {
 	       .name = "pm8916_wcd_analog_pdm_tx",
@@ -781,7 +764,6 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = {
 			   .channels_min = 1,
 			   .channels_max = 4,
 			   },
-	       .ops = &pm8916_wcd_analog_dai_ops,
 	       },
 };
 
-- 
cgit v1.2.3


From 0d7820d0918efa6c2a53acdfc47cbfc3b48df727 Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Thu, 10 Aug 2017 17:13:19 +0200
Subject: ASoC: rsnd: Delete an error message for a failed memory allocation in
 three functions

Omit an extra message for a memory allocation failure in these functions.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/adg.c  | 4 +---
 sound/soc/sh/rcar/core.c | 4 +---
 sound/soc/sh/rcar/gen.c  | 4 +---
 3 files changed, 3 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 5b5389e5b92b..938baff86ef2 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -586,10 +586,8 @@ int rsnd_adg_probe(struct rsnd_priv *priv)
 	int ret;
 
 	adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
-	if (!adg) {
-		dev_err(dev, "ADG allocate failed\n");
+	if (!adg)
 		return -ENOMEM;
-	}
 
 	ret = rsnd_mod_init(priv, &adg->mod, &adg_ops,
 		      NULL, NULL, 0, 0);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 2c12ad2b2b34..650cc28b0fb6 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1401,10 +1401,8 @@ static int rsnd_probe(struct platform_device *pdev)
 	 *	init priv data
 	 */
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
-	if (!priv) {
-		dev_err(dev, "priv allocate failed\n");
+	if (!priv)
 		return -ENODEV;
-	}
 
 	priv->pdev	= pdev;
 	priv->flags	= (unsigned long)of_device_get_match_data(dev);
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index ee00e3516911..f04c4100043a 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -406,10 +406,8 @@ int rsnd_gen_probe(struct rsnd_priv *priv)
 	int ret;
 
 	gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL);
-	if (!gen) {
-		dev_err(dev, "GEN allocate failed\n");
+	if (!gen)
 		return -ENOMEM;
-	}
 
 	priv->gen = gen;
 
-- 
cgit v1.2.3


From a0ac441152238c9b474bafa46940511d9b2e9c7e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 8 Aug 2017 06:17:47 +0000
Subject: ASoC: soc-core: add snd_soc_rtdcom_xxx()

Current snd_soc_pcm_runtime has platform / codec pointers, and we could
use these specific pointer. But these will be replaced to more generic
"component" soon, and will need more generic method to get each
connected component pointer from rtd.

This patch adds new snd_soc_rtdcom_xxx() to connect/disconnect
component to rtd. It means same as previous "platform" / "codec"
pointer style, but more generic.
We can find necessary component pointer from rtd by using component
driver name on snd_soc_rtdcom_lookup().

Here, the reason why it uses "driver name" is that "component name"
was created by fmt_single_name() and difficult to use it from driver.
Driver of course knows its "driver name", thus, using it is more easy.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 13 +++++++++++++
 sound/soc/soc-core.c | 50 ++++++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 63 insertions(+)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index aad3f14966f1..87dbe39797ca 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -886,6 +886,18 @@ struct snd_soc_component {
 #endif
 };
 
+struct snd_soc_rtdcom_list {
+	struct snd_soc_component *component;
+	struct list_head list; /* rtd::component_list */
+};
+struct snd_soc_component*
+snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd,
+		       const char *driver_name);
+#define for_each_rtdcom(rtd, rtdcom) \
+	list_for_each_entry(rtdcom, &(rtd)->component_list, list)
+#define for_each_rtdcom_safe(rtd, rtdcom1, rtdcom2) \
+	list_for_each_entry_safe(rtdcom1, rtdcom2, &(rtd)->component_list, list)
+
 /* SoC Audio Codec device */
 struct snd_soc_codec {
 	struct device *dev;
@@ -1240,6 +1252,7 @@ struct snd_soc_pcm_runtime {
 
 	unsigned int num; /* 0-based and monotonic increasing */
 	struct list_head list; /* rtd list of the soc card */
+	struct list_head component_list; /* list of connected components */
 
 	/* bit field */
 	unsigned int dev_registered:1;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 07d708498500..c2462e0f0f0d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -551,6 +551,54 @@ static inline void snd_soc_debugfs_exit(void)
 
 #endif
 
+static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd,
+			      struct snd_soc_component *component)
+{
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_rtdcom_list *new_rtdcom;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		/* already connected */
+		if (rtdcom->component == component)
+			return 0;
+	}
+
+	new_rtdcom = kmalloc(sizeof(*new_rtdcom), GFP_KERNEL);
+	if (!new_rtdcom)
+		return -ENOMEM;
+
+	new_rtdcom->component = component;
+	INIT_LIST_HEAD(&new_rtdcom->list);
+
+	list_add_tail(&new_rtdcom->list, &rtd->component_list);
+
+	return 0;
+}
+
+static void snd_soc_rtdcom_del_all(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_rtdcom_list *rtdcom1, *rtdcom2;
+
+	for_each_rtdcom_safe(rtd, rtdcom1, rtdcom2)
+		kfree(rtdcom1);
+
+	INIT_LIST_HEAD(&rtd->component_list);
+}
+
+struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd,
+						const char *driver_name)
+{
+	struct snd_soc_rtdcom_list *rtdcom;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		if ((rtdcom->component->driver->name == driver_name) ||
+		    strcmp(rtdcom->component->driver->name, driver_name) == 0)
+			return rtdcom->component;
+	}
+
+	return NULL;
+}
+
 struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
 		const char *dai_link, int stream)
 {
@@ -575,6 +623,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
 	if (!rtd)
 		return NULL;
 
+	INIT_LIST_HEAD(&rtd->component_list);
 	rtd->card = card;
 	rtd->dai_link = dai_link;
 	rtd->codec_dais = kzalloc(sizeof(struct snd_soc_dai *) *
@@ -592,6 +641,7 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd)
 {
 	if (rtd && rtd->codec_dais)
 		kfree(rtd->codec_dais);
+	snd_soc_rtdcom_del_all(rtd);
 	kfree(rtd);
 }
 
-- 
cgit v1.2.3


From 90be711e23f7c7ee7b3d6a6e5aa7ee9bab321f2e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 8 Aug 2017 06:18:10 +0000
Subject: ASoC: use snd_soc_rtdcom_add() and convert to consistent operation

Basically, current ALSA SoC framework is based on CPU/Codec/Platform,
but its operation doesn't have consistent.
Thus, source code was unreadable, and difficult to understand.
This patch connects each component (= CPU/Codec/Platform) to rtd by
using snd_soc_rtdcom_add(), and convert uneven operations to consistent
operation.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  2 +-
 sound/soc/soc-core.c | 61 +++++++++++++++++++++++-----------------------------
 sound/soc/soc-pcm.c  | 38 ++++++++++++++++----------------
 3 files changed, 47 insertions(+), 54 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 87dbe39797ca..82e078151787 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1237,7 +1237,7 @@ struct snd_soc_pcm_runtime {
 	struct snd_pcm *pcm;
 	struct snd_compr *compr;
 	struct snd_soc_codec *codec;
-	struct snd_soc_platform *platform;
+	struct snd_soc_platform *platform; /* will be removed */
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c2462e0f0f0d..bbbe003fc0af 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1100,6 +1100,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai_link_component *codecs = dai_link->codecs;
 	struct snd_soc_dai_link_component cpu_dai_component;
+	struct snd_soc_component *component;
 	struct snd_soc_dai **codec_dais;
 	struct snd_soc_platform *platform;
 	struct device_node *platform_of_node;
@@ -1127,6 +1128,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 			dai_link->cpu_dai_name);
 		goto _err_defer;
 	}
+	snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component);
 
 	rtd->num_codecs = dai_link->num_codecs;
 
@@ -1139,6 +1141,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 				codecs[i].dai_name);
 			goto _err_defer;
 		}
+		snd_soc_rtdcom_add(rtd, codec_dais[i]->component);
 	}
 
 	/* Single codec links expect codec and codec_dai in runtime data */
@@ -1150,6 +1153,23 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	if (!platform_name && !dai_link->platform_of_node)
 		platform_name = "snd-soc-dummy";
 
+	/* find one from the set of registered platforms */
+	list_for_each_entry(component, &component_list, list) {
+		platform_of_node = component->dev->of_node;
+		if (!platform_of_node && component->dev->parent->of_node)
+			platform_of_node = component->dev->parent->of_node;
+
+		if (dai_link->platform_of_node) {
+			if (platform_of_node != dai_link->platform_of_node)
+				continue;
+		} else {
+			if (strcmp(component->name, platform_name))
+				continue;
+		}
+
+		snd_soc_rtdcom_add(rtd, component);
+	}
+
 	/* find one from the set of registered platforms */
 	list_for_each_entry(platform, &platform_list, list) {
 		platform_of_node = platform->dev->of_node;
@@ -1235,27 +1255,15 @@ static void soc_remove_link_dais(struct snd_soc_card *card,
 static void soc_remove_link_components(struct snd_soc_card *card,
 	struct snd_soc_pcm_runtime *rtd, int order)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_component *component;
-	int i;
+	struct snd_soc_rtdcom_list *rtdcom;
 
-	/* remove the platform */
-	if (platform && platform->component.driver->remove_order == order)
-		soc_remove_component(&platform->component);
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
 
-	/* remove the CODEC-side CODEC */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		component = rtd->codec_dais[i]->component;
 		if (component->driver->remove_order == order)
 			soc_remove_component(component);
 	}
-
-	/* remove any CPU-side CODEC */
-	if (cpu_dai) {
-		if (cpu_dai->component->driver->remove_order == order)
-			soc_remove_component(cpu_dai->component);
-	}
 }
 
 static void soc_remove_dai_links(struct snd_soc_card *card)
@@ -1607,21 +1615,13 @@ static int soc_probe_link_components(struct snd_soc_card *card,
 			struct snd_soc_pcm_runtime *rtd,
 				     int order)
 {
-	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_component *component;
-	int i, ret;
+	struct snd_soc_rtdcom_list *rtdcom;
+	int ret;
 
-	/* probe the CPU-side component, if it is a CODEC */
-	component = rtd->cpu_dai->component;
-	if (component->driver->probe_order == order) {
-		ret = soc_probe_component(card, component);
-		if (ret < 0)
-			return ret;
-	}
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
 
-	/* probe the CODEC-side components */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		component = rtd->codec_dais[i]->component;
 		if (component->driver->probe_order == order) {
 			ret = soc_probe_component(card, component);
 			if (ret < 0)
@@ -1629,13 +1629,6 @@ static int soc_probe_link_components(struct snd_soc_card *card,
 		}
 	}
 
-	/* probe the platform */
-	if (platform->component.driver->probe_order == order) {
-		ret = soc_probe_component(card, &platform->component);
-		if (ret < 0)
-			return ret;
-	}
-
 	return 0;
 }
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index dcc5ece08668..aa4c02de9c61 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -450,6 +450,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_component *component;
+	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	const char *codec_dai_name = "multicodec";
@@ -458,10 +460,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	pinctrl_pm_select_default_state(cpu_dai->dev);
 	for (i = 0; i < rtd->num_codecs; i++)
 		pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev);
-	pm_runtime_get_sync(cpu_dai->dev);
-	for (i = 0; i < rtd->num_codecs; i++)
-		pm_runtime_get_sync(rtd->codec_dais[i]->dev);
-	pm_runtime_get_sync(platform->dev);
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		pm_runtime_get_sync(component->dev);
+	}
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
@@ -599,15 +603,13 @@ platform_err:
 out:
 	mutex_unlock(&rtd->pcm_mutex);
 
-	pm_runtime_mark_last_busy(platform->dev);
-	pm_runtime_put_autosuspend(platform->dev);
-	for (i = 0; i < rtd->num_codecs; i++) {
-		pm_runtime_mark_last_busy(rtd->codec_dais[i]->dev);
-		pm_runtime_put_autosuspend(rtd->codec_dais[i]->dev);
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		pm_runtime_mark_last_busy(component->dev);
+		pm_runtime_put_autosuspend(component->dev);
 	}
 
-	pm_runtime_mark_last_busy(cpu_dai->dev);
-	pm_runtime_put_autosuspend(cpu_dai->dev);
 	for (i = 0; i < rtd->num_codecs; i++) {
 		if (!rtd->codec_dais[i]->active)
 			pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
@@ -655,6 +657,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_component *component;
+	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	int i;
@@ -711,17 +715,13 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 
 	mutex_unlock(&rtd->pcm_mutex);
 
-	pm_runtime_mark_last_busy(platform->dev);
-	pm_runtime_put_autosuspend(platform->dev);
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		pm_runtime_mark_last_busy(rtd->codec_dais[i]->dev);
-		pm_runtime_put_autosuspend(rtd->codec_dais[i]->dev);
+		pm_runtime_mark_last_busy(component->dev);
+		pm_runtime_put_autosuspend(component->dev);
 	}
 
-	pm_runtime_mark_last_busy(cpu_dai->dev);
-	pm_runtime_put_autosuspend(cpu_dai->dev);
-
 	for (i = 0; i < rtd->num_codecs; i++) {
 		if (!rtd->codec_dais[i]->active)
 			pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
-- 
cgit v1.2.3


From 4428ffa1fe66a9e3cf68bba3ad135f17be4ed674 Mon Sep 17 00:00:00 2001
From: Sapthagiri Baratam
Date: Mon, 14 Aug 2017 15:46:24 +0100
Subject: arizona: anc: Correct setting of FCx_MIC_MODE_SEL

The mask rather than the shift is accidentally passed to the
SOC_ENUM_SINGLE macro which results in the wrong bits being set for this
control, change to using the shift to correct this.

Signed-off-by: Sapthagiri Baratam <sapthagiri.baratam@cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/arizona.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 0a734d910850..a1149f6a8450 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -710,7 +710,7 @@ const struct soc_enum arizona_anc_input_src[] = {
 			ARRAY_SIZE(arizona_anc_input_src_text),
 			arizona_anc_input_src_text),
 	SOC_ENUM_SINGLE(ARIZONA_FCL_ADC_REFORMATTER_CONTROL,
-			ARIZONA_FCL_MIC_MODE_SEL,
+			ARIZONA_FCL_MIC_MODE_SEL_SHIFT,
 			ARRAY_SIZE(arizona_anc_channel_src_text),
 			arizona_anc_channel_src_text),
 	SOC_ENUM_SINGLE(ARIZONA_ANC_SRC,
@@ -718,7 +718,7 @@ const struct soc_enum arizona_anc_input_src[] = {
 			ARRAY_SIZE(arizona_anc_input_src_text),
 			arizona_anc_input_src_text),
 	SOC_ENUM_SINGLE(ARIZONA_FCR_ADC_REFORMATTER_CONTROL,
-			ARIZONA_FCR_MIC_MODE_SEL,
+			ARIZONA_FCR_MIC_MODE_SEL_SHIFT,
 			ARRAY_SIZE(arizona_anc_channel_src_text),
 			arizona_anc_channel_src_text),
 };
-- 
cgit v1.2.3


From 9ab51b9469e6a63dbf4385bdd02704e87b65a039 Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Sun, 13 Aug 2017 16:00:33 +0100
Subject: ASoC: max98926: make max98926_spk_tlv and max98926_current_tlv static

max98926_spk_tlv and max98926_current_tlv are local to the source and do
not need to be in global scope, so make them static.

Cleans up sparse warnings:
symbol 'max98926_spk_tlv' was not declared. Should it be static?
symbol 'max98926_current_tlv' was not declared. Should it be static?

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98926.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
index f16145d8ee6b..13d2708657e1 100644
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -213,8 +213,8 @@ static bool max98926_readable_register(struct device *dev, unsigned int reg)
 	}
 };
 
-DECLARE_TLV_DB_SCALE(max98926_spk_tlv, -600, 100, 0);
-DECLARE_TLV_DB_RANGE(max98926_current_tlv,
+static DECLARE_TLV_DB_SCALE(max98926_spk_tlv, -600, 100, 0);
+static DECLARE_TLV_DB_RANGE(max98926_current_tlv,
 	0, 11, TLV_DB_SCALE_ITEM(20, 20, 0),
 	12, 15, TLV_DB_SCALE_ITEM(320, 40, 0),
 );
-- 
cgit v1.2.3


From e5e8e6182a3b9514cde41d957684b59ee8f6cf21 Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Sun, 13 Aug 2017 16:08:03 +0100
Subject: ASoC: rt5514: make array rt5514_dai static

The array rt5514_dai is local to the source and does not need to be in
global scope, so make it static.

Cleans up sparse warnings:
symbol 'rt5514_dai' was not declared. Should it be static?

warning: symbol 'rt5514_dai' was not declared. Should it be static?

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 0eabe569ab73..a694ce9fed86 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -1066,7 +1066,7 @@ static const struct snd_soc_dai_ops rt5514_aif_dai_ops = {
 	.set_tdm_slot = rt5514_set_tdm_slot,
 };
 
-struct snd_soc_dai_driver rt5514_dai[] = {
+static struct snd_soc_dai_driver rt5514_dai[] = {
 	{
 		.name = "rt5514-aif1",
 		.id = 0,
-- 
cgit v1.2.3


From def4b1243e56774fdc47de9292111a344f14e88c Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Sun, 13 Aug 2017 16:28:08 +0100
Subject: ASoC: zx_aud96p22: make array aud96p22_dt_ids static

The array aud96p22_dt_ids is local to the source and does not need to be
in global scope, so make it static.

Cleans up sparse warning:
symbol 'aud96p22_dt_ids' was not declared. Should it be static?

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/zx_aud96p22.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c
index 032fb7cf6cbd..bc23ce1a30a1 100644
--- a/sound/soc/codecs/zx_aud96p22.c
+++ b/sound/soc/codecs/zx_aud96p22.c
@@ -382,7 +382,7 @@ static int aud96p22_i2c_remove(struct i2c_client *i2c)
 	return 0;
 }
 
-const struct of_device_id aud96p22_dt_ids[] = {
+static const struct of_device_id aud96p22_dt_ids[] = {
 	{ .compatible = "zte,zx-aud96p22", },
 	{ }
 };
-- 
cgit v1.2.3


From fa99125e2acca4e85de57cff55aea9fc7cb2930a Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Sun, 13 Aug 2017 16:37:15 +0100
Subject: ASoC: samsung: make tm2_dapm_widgets and tm2_pm_ops static

tm2_dapm_widgets and tm2_pm_ops are local to the source and do not need
to be in global scope, so make them static.

Cleans up sparse warnings:
symbol 'tm2_dapm_widgets' was not declared. Should it be static?
symbol 'tm2_pm_ops' was not declared. Should it be static?

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/tm2_wm5110.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index 24cc9d63ce87..68698f3d72f9 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -318,7 +318,7 @@ static const struct snd_kcontrol_new tm2_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
 };
 
-const struct snd_soc_dapm_widget tm2_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget tm2_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("HP", NULL),
 	SND_SOC_DAPM_SPK("SPK", NULL),
 	SND_SOC_DAPM_SPK("RCV", NULL),
@@ -521,7 +521,7 @@ static void tm2_pm_complete(struct device *dev)
 	tm2_start_sysclk(card);
 }
 
-const struct dev_pm_ops tm2_pm_ops = {
+static const struct dev_pm_ops tm2_pm_ops = {
 	.prepare	= tm2_pm_prepare,
 	.suspend	= snd_soc_suspend,
 	.resume		= snd_soc_resume,
-- 
cgit v1.2.3


From 0cc798651a7796712a7bee1f3c865e82967d933f Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:31 +0530
Subject: ASoC: samsung: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/idma.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index 3e408158625d..5c59c95c5ade 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -325,7 +325,7 @@ static int idma_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static struct snd_pcm_ops idma_ops = {
+static const struct snd_pcm_ops idma_ops = {
 	.open		= idma_open,
 	.close		= idma_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From d92dd70e9a47cce7d51bda584e7910efd0c4ba20 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:47 +0530
Subject: ASoC: samsung: make snd_soc_platform_driver const

Make this const as it is only passed as the 2nd argument to the function
devm_snd_soc_register_platform, which is of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/idma.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index 5c59c95c5ade..a635df61f928 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -399,7 +399,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr)
 }
 EXPORT_SYMBOL_GPL(idma_reg_addr_init);
 
-static struct snd_soc_platform_driver asoc_idma_platform = {
+static const struct snd_soc_platform_driver asoc_idma_platform = {
 	.ops = &idma_ops,
 	.pcm_new = idma_new,
 	.pcm_free = idma_free,
-- 
cgit v1.2.3


From c5755fbc0702ddb38d2b476a49327b59d4d75a00 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Mon, 14 Aug 2017 09:46:59 +0800
Subject: ASoC: rt5663: Fine tune for the headphone output pop sound

The patch fine tunes for the headphone output pop sound.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 21 ++++++++++++++++-----
 1 file changed, 16 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index be93231b0f58..fdc2bbcb67c2 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -1565,6 +1565,10 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 		val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & 0x0003;
 		dev_dbg(codec->dev, "%s val = %d\n", __func__, val);
 
+		snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1,
+			RT5663_OSW_HP_L_MASK | RT5663_OSW_HP_R_MASK,
+			RT5663_OSW_HP_L_EN | RT5663_OSW_HP_R_EN);
+
 		switch (val) {
 		case 1:
 		case 2:
@@ -2073,6 +2077,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 			snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7766);
 			snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa);
 			snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777);
+			snd_soc_update_bits(codec, RT5663_STO_DRE_1, 0x8000,
+				0x8000);
 			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000,
 				0x3000);
 		}
@@ -2105,14 +2111,19 @@ static int rt5663_charge_pump_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		if (rt5663->codec_ver == CODEC_VER_0)
-			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x003b,
-				0x003b);
+		if (rt5663->codec_ver == CODEC_VER_0) {
+			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030,
+				0x0030);
+			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003,
+				0x0003);
+		}
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
-		if (rt5663->codec_ver == CODEC_VER_0)
-			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x003b, 0);
+		if (rt5663->codec_ver == CODEC_VER_0) {
+			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0003, 0);
+			snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0);
+		}
 		break;
 
 	default:
-- 
cgit v1.2.3


From e731a870153271c861f06dc09e6acb5781b973f3 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:24 +0530
Subject: ASoC: au1x: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/au1x/dbdma2.c | 2 +-
 sound/soc/au1x/dma.c    | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index b5d1caa04d8e..6a035ca0f521 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -304,7 +304,7 @@ static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static struct snd_pcm_ops au1xpsc_pcm_ops = {
+static const struct snd_pcm_ops au1xpsc_pcm_ops = {
 	.open		= au1xpsc_pcm_open,
 	.close		= au1xpsc_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
index fcf5a9adde81..19457e2b351e 100644
--- a/sound/soc/au1x/dma.c
+++ b/sound/soc/au1x/dma.c
@@ -277,7 +277,7 @@ static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss)
 	return bytes_to_frames(ss->runtime, location);
 }
 
-static struct snd_pcm_ops alchemy_pcm_ops = {
+static const struct snd_pcm_ops alchemy_pcm_ops = {
 	.open			= alchemy_pcm_open,
 	.close			= alchemy_pcm_close,
 	.ioctl			= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From e76b3f41a3c39fae342fe4db530c3917199cfa96 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:25 +0530
Subject: ASoC: blackfin: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +-
 sound/soc/blackfin/bf5xx-i2s-pcm.c  | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 913e29275f4e..8c1d1983b8f9 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -308,7 +308,7 @@ static	int bf5xx_pcm_copy_user(struct snd_pcm_substream *substream,
 }
 #endif
 
-static struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
+static const struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
 	.open		= bf5xx_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 470d99abf6f6..51cae76f14e6 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -318,7 +318,7 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
+static const struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
 	.open		= bf5xx_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
-- 
cgit v1.2.3


From b6ed072028705620ea2937ddafd77a202df034af Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:26 +0530
Subject: ASoC: fsl: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_dma.c     | 2 +-
 sound/soc/fsl/imx-pcm-fiq.c | 2 +-
 sound/soc/fsl/mpc5200_dma.c | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index ed8ea002902d..979ed1b3ecc4 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -870,7 +870,7 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
 	return NULL;
 }
 
-static struct snd_pcm_ops fsl_dma_ops = {
+static const struct snd_pcm_ops fsl_dma_ops = {
 	.open   	= fsl_dma_open,
 	.close  	= fsl_dma_close,
 	.ioctl  	= snd_pcm_lib_ioctl,
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 92410f7ca1fa..9ce37616ef5d 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -227,7 +227,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static struct snd_pcm_ops imx_pcm_ops = {
+static const struct snd_pcm_ops imx_pcm_ops = {
 	.open		= snd_imx_open,
 	.close		= snd_imx_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 1f7e70bfbd55..66042ce86edc 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -287,7 +287,7 @@ psc_dma_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_pcm_ops psc_dma_ops = {
+static const struct snd_pcm_ops psc_dma_ops = {
 	.open		= psc_dma_open,
 	.close		= psc_dma_close,
 	.hw_free	= psc_dma_hw_free,
-- 
cgit v1.2.3


From 779238e28e42ff49f314df217de146b14d72c582 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:42 +0530
Subject: ASoC: fsl: make snd_soc_platform_driver const

Make these const as they are only passed as the 2nd argument to the function
snd_soc_register_platform, which is of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/imx-pcm-fiq.c | 2 +-
 sound/soc/fsl/mpc5200_dma.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 9ce37616ef5d..aef1f7819c40 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -341,7 +341,7 @@ static void imx_pcm_fiq_free(struct snd_pcm *pcm)
 	imx_pcm_free(pcm);
 }
 
-static struct snd_soc_platform_driver imx_soc_platform_fiq = {
+static const struct snd_soc_platform_driver imx_soc_platform_fiq = {
 	.ops		= &imx_pcm_ops,
 	.pcm_new	= imx_pcm_fiq_new,
 	.pcm_free	= imx_pcm_fiq_free,
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 66042ce86edc..e63029f1aabc 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -356,7 +356,7 @@ static void psc_dma_free(struct snd_pcm *pcm)
 	}
 }
 
-static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
+static const struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
 	.ops		= &psc_dma_ops,
 	.pcm_new	= &psc_dma_new,
 	.pcm_free	= &psc_dma_free,
-- 
cgit v1.2.3


From 83823c43c25a62b529f9dbad48c182b3d1fdddd1 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:43 +0530
Subject: ASoC: nuc900: make snd_soc_platform_driver const

Make this const as it is only passed as the 2nd argument to the
function devm_snd_soc_register_platform, which is of type const.
Done using Coccinelle

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/nuc900/nuc900-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 2cca055fd806..cd0486cad7f5 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -299,7 +299,7 @@ static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static struct snd_soc_platform_driver nuc900_soc_platform = {
+static const struct snd_soc_platform_driver nuc900_soc_platform = {
 	.ops		= &nuc900_dma_ops,
 	.pcm_new	= nuc900_dma_new,
 };
-- 
cgit v1.2.3


From 3a7f221723a6f0032a7d908f3f91ab8604f0311f Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:29 +0530
Subject: ASoC: omap: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/omap/omap-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 94e9ff791f3a..e2f12450fb66 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -162,7 +162,7 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream,
 			   runtime->dma_addr, runtime->dma_bytes);
 }
 
-static struct snd_pcm_ops omap_pcm_ops = {
+static const struct snd_pcm_ops omap_pcm_ops = {
 	.open		= omap_pcm_open,
 	.close		= snd_dmaengine_pcm_close_release_chan,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From 27bfda02662152f5e0d17731063cfcb342904b58 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:44 +0530
Subject: ASoC: omap: make snd_soc_platform_driver const

Make this const as it is only passed as the 2nd argument to the
function devm_snd_soc_register_platform, which is of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/omap/omap-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e2f12450fb66..aca2c43d0f03 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -243,7 +243,7 @@ out:
 	return ret;
 }
 
-static struct snd_soc_platform_driver omap_soc_platform = {
+static const struct snd_soc_platform_driver omap_soc_platform = {
 	.ops		= &omap_pcm_ops,
 	.pcm_new	= omap_pcm_new,
 	.pcm_free	= omap_pcm_free_dma_buffers,
-- 
cgit v1.2.3


From 3d882b0ec6a50c5adff8e94a69d284543460fbe6 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:45 +0530
Subject: ASoC: pxa: make snd_soc_platform_driver const

Make these const as they are only passed as the 2nd argument to the
function devm_snd_soc_register_platform, which is of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/pxa/mmp-pcm.c    | 2 +-
 sound/soc/pxa/pxa2xx-pcm.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 5b5f1a442891..38d599949ee4 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -211,7 +211,7 @@ err:
 	return ret;
 }
 
-static struct snd_soc_platform_driver mmp_soc_platform = {
+static const struct snd_soc_platform_driver mmp_soc_platform = {
 	.ops		= &mmp_pcm_ops,
 	.pcm_new	= mmp_pcm_new,
 	.pcm_free	= mmp_pcm_free_dma_buffers,
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index b51d7a0755d5..636895a778cb 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -84,7 +84,7 @@ static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
-static struct snd_soc_platform_driver pxa2xx_soc_platform = {
+static const struct snd_soc_platform_driver pxa2xx_soc_platform = {
 	.ops		= &pxa2xx_pcm_ops,
 	.pcm_new	= pxa2xx_soc_pcm_new,
 	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
-- 
cgit v1.2.3


From d7622fa715f20dbc7c1ab0dbdd714be5a2b4c3bd Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:30 +0530
Subject: ASoC: pxa: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/pxa/mmp-pcm.c    | 2 +-
 sound/soc/pxa/pxa2xx-pcm.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 38d599949ee4..624d9bd5dadd 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -131,7 +131,7 @@ static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
 		vma->vm_end - vma->vm_start, vma->vm_page_prot);
 }
 
-static struct snd_pcm_ops mmp_pcm_ops = {
+static const struct snd_pcm_ops mmp_pcm_ops = {
 	.open		= mmp_pcm_open,
 	.close		= snd_dmaengine_pcm_close_release_chan,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 636895a778cb..e64958d8bff0 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -45,7 +45,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static struct snd_pcm_ops pxa2xx_pcm_ops = {
+static const struct snd_pcm_ops pxa2xx_pcm_ops = {
 	.open		= __pxa2xx_pcm_open,
 	.close		= __pxa2xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From 50d50c33d29a868bd58e0f320c19b9e9e7084a2a Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:46 +0530
Subject: ASoC: qcom: make snd_soc_platform_driver const

Make this const as it is only passed as the 2nd argument to the
function devm_snd_soc_register_platform, which is of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/lpass-platform.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 7aabf08de3d4..fb3576af7911 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -557,7 +557,7 @@ static void lpass_platform_pcm_free(struct snd_pcm *pcm)
 	}
 }
 
-static struct snd_soc_platform_driver lpass_platform_driver = {
+static const struct snd_soc_platform_driver lpass_platform_driver = {
 	.pcm_new	= lpass_platform_pcm_new,
 	.pcm_free	= lpass_platform_pcm_free,
 	.ops		= &lpass_platform_pcm_ops,
-- 
cgit v1.2.3


From 517725545179be8d32730844f1619ee31ef04bf3 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:48 +0530
Subject: ASoC: sh: make snd_soc_platform_driver const

Make these const as they are either passed as the 2nd argument to
the function devm_snd_soc_register_platform or snd_soc_register_platform,
and the arguments are of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/dma-sh7760.c | 2 +-
 sound/soc/sh/fsi.c        | 2 +-
 sound/soc/sh/rcar/core.c  | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 8fad4441c87d..81c743322445 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -320,7 +320,7 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static struct snd_soc_platform_driver sh7760_soc_platform = {
+static const struct snd_soc_platform_driver sh7760_soc_platform = {
 	.ops		= &camelot_pcm_ops,
 	.pcm_new	= camelot_pcm_new,
 };
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 7c4bdd82bb95..90021b8a799a 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1818,7 +1818,7 @@ static struct snd_soc_dai_driver fsi_soc_dai[] = {
 	},
 };
 
-static struct snd_soc_platform_driver fsi_soc_platform = {
+static const struct snd_soc_platform_driver fsi_soc_platform = {
 	.ops		= &fsi_pcm_ops,
 	.pcm_new	= fsi_pcm_new,
 };
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 3f2ced26ed37..e50f7e7981c8 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1339,7 +1339,7 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd)
 		PREALLOC_BUFFER, PREALLOC_BUFFER_MAX);
 }
 
-static struct snd_soc_platform_driver rsnd_soc_platform = {
+static const struct snd_soc_platform_driver rsnd_soc_platform = {
 	.ops		= &rsnd_pcm_ops,
 	.pcm_new	= rsnd_pcm_new,
 };
-- 
cgit v1.2.3


From 42271d951c1f9a6a3ae0392d245256d973b2fc58 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:49 +0530
Subject: ASoC: txx9: make snd_soc_platform_driver const

Make this const as it is only passed as the 2nd argument to the function
devm_snd_soc_register_platform, which is of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/txx9/txx9aclc.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 7912bf09dc4d..7df95dfa8557 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -403,7 +403,7 @@ static int txx9aclc_pcm_remove(struct snd_soc_platform *platform)
 	return 0;
 }
 
-static struct snd_soc_platform_driver txx9aclc_soc_platform = {
+static const struct snd_soc_platform_driver txx9aclc_soc_platform = {
 	.probe		= txx9aclc_pcm_probe,
 	.remove		= txx9aclc_pcm_remove,
 	.ops		= &txx9aclc_pcm_ops,
-- 
cgit v1.2.3


From 5144468efb076b1859477de19257dd7ec9c24ed3 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:33 +0530
Subject: ASoC: txx9: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/txx9/txx9aclc.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 7df95dfa8557..a2bb68fea5a3 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -271,7 +271,7 @@ static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static struct snd_pcm_ops txx9aclc_pcm_ops = {
+static const struct snd_pcm_ops txx9aclc_pcm_ops = {
 	.open		= txx9aclc_pcm_open,
 	.close		= txx9aclc_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From c9f3d3441311792a0beb410af74e1aad4c3f715d Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:50 +0530
Subject: ASoC: soc-utils: make snd_soc_platform_driver const

Make this const as it is only passed as the 2nd argument to the function
snd_soc_register_platform, which is of type const.
Done using Coccinelle

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-utils.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 644d9a9ebfbc..e30aacbcfc29 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -284,7 +284,7 @@ static const struct snd_pcm_ops dummy_dma_ops = {
 	.ioctl		= snd_pcm_lib_ioctl,
 };
 
-static struct snd_soc_platform_driver dummy_platform = {
+static const struct snd_soc_platform_driver dummy_platform = {
 	.ops = &dummy_dma_ops,
 };
 
-- 
cgit v1.2.3


From 0ed6f15701efa0d62b4556900cd67a726578389e Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:40 +0530
Subject: ASoC: codecs: make snd_soc_platform_driver const

Make these const as they are either passed as the 2nd argument to the
function devm_snd_soc_register_platform or snd_soc_register_platform,
and the arguments are of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs47l24.c    | 2 +-
 sound/soc/codecs/rt5514-spi.c | 2 +-
 sound/soc/codecs/wm5102.c     | 2 +-
 sound/soc/codecs/wm5110.c     | 2 +-
 4 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index d323caa9c816..505dbc9d73cf 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -1213,7 +1213,7 @@ static struct snd_compr_ops cs47l24_compr_ops = {
 	.copy = wm_adsp_compr_copy,
 };
 
-static struct snd_soc_platform_driver cs47l24_compr_platform = {
+static const struct snd_soc_platform_driver cs47l24_compr_platform = {
 	.compr_ops = &cs47l24_compr_ops,
 };
 
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 7ed62e8c80b4..87a587fa90fe 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -260,7 +260,7 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform)
 	return 0;
 }
 
-static struct snd_soc_platform_driver rt5514_spi_platform = {
+static const struct snd_soc_platform_driver rt5514_spi_platform = {
 	.probe = rt5514_spi_pcm_probe,
 	.ops = &rt5514_spi_pcm_ops,
 };
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 1fe358e6be61..f5006923be2e 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -2027,7 +2027,7 @@ static struct snd_compr_ops wm5102_compr_ops = {
 	.copy = wm_adsp_compr_copy,
 };
 
-static struct snd_soc_platform_driver wm5102_compr_platform = {
+static const struct snd_soc_platform_driver wm5102_compr_platform = {
 	.compr_ops = &wm5102_compr_ops,
 };
 
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 1bc942152eff..d6fae1397ce2 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2382,7 +2382,7 @@ static struct snd_compr_ops wm5110_compr_ops = {
 	.copy = wm_adsp_compr_copy,
 };
 
-static struct snd_soc_platform_driver wm5110_compr_platform = {
+static const struct snd_soc_platform_driver wm5110_compr_platform = {
 	.compr_ops = &wm5110_compr_ops,
 };
 
-- 
cgit v1.2.3


From 80cc4df8b46e91c28a6d39dd0ba76bd9c83c18ce Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Mon, 14 Aug 2017 17:08:41 +0530
Subject: ASoC: Intel: make snd_soc_platform_driver const

Make these const as they are only passed as the 2nd argument to the
function snd_soc_register_platform, which is of type const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +-
 sound/soc/intel/baytrail/sst-baytrail-pcm.c  | 2 +-
 sound/soc/intel/haswell/sst-haswell-pcm.c    | 2 +-
 sound/soc/intel/skylake/skl-pcm.c            | 2 +-
 4 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 49c7b88b5bdc..b272df5ce0e8 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -705,7 +705,7 @@ static int sst_soc_probe(struct snd_soc_platform *platform)
 	return sst_dsp_init_v2_dpcm(platform);
 }
 
-static struct snd_soc_platform_driver sst_soc_platform_drv  = {
+static const struct snd_soc_platform_driver sst_soc_platform_drv  = {
 	.probe		= sst_soc_probe,
 	.ops		= &sst_platform_ops,
 	.compr_ops	= &sst_platform_compr_ops,
diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
index 4765ad474544..84cb568c3ad6 100644
--- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c
+++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
@@ -395,7 +395,7 @@ static int sst_byt_pcm_remove(struct snd_soc_platform *platform)
 	return 0;
 }
 
-static struct snd_soc_platform_driver byt_soc_platform = {
+static const struct snd_soc_platform_driver byt_soc_platform = {
 	.probe		= sst_byt_pcm_probe,
 	.remove		= sst_byt_pcm_remove,
 	.ops		= &sst_byt_pcm_ops,
diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c
index 9e4094e2c6e3..c044400540ec 100644
--- a/sound/soc/intel/haswell/sst-haswell-pcm.c
+++ b/sound/soc/intel/haswell/sst-haswell-pcm.c
@@ -1135,7 +1135,7 @@ static int hsw_pcm_remove(struct snd_soc_platform *platform)
 	return 0;
 }
 
-static struct snd_soc_platform_driver hsw_soc_platform = {
+static const struct snd_soc_platform_driver hsw_soc_platform = {
 	.probe		= hsw_pcm_probe,
 	.remove		= hsw_pcm_remove,
 	.ops		= &hsw_pcm_ops,
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index debdaac2c3a2..e98d8252a026 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -1310,7 +1310,7 @@ static int skl_platform_soc_probe(struct snd_soc_platform *platform)
 
 	return 0;
 }
-static struct snd_soc_platform_driver skl_platform_drv  = {
+static const struct snd_soc_platform_driver skl_platform_drv  = {
 	.probe		= skl_platform_soc_probe,
 	.ops		= &skl_platform_ops,
 	.pcm_new	= skl_pcm_new,
-- 
cgit v1.2.3


From f2c402ba55fe7cc3924de683ac5eef99c75eab04 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:27 +0530
Subject: ASoC: intel: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
index 84cb568c3ad6..c54529320f07 100644
--- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c
+++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
@@ -309,7 +309,7 @@ static int sst_byt_pcm_mmap(struct snd_pcm_substream *substream,
 	return snd_pcm_lib_default_mmap(substream, vma);
 }
 
-static struct snd_pcm_ops sst_byt_pcm_ops = {
+static const struct snd_pcm_ops sst_byt_pcm_ops = {
 	.open		= sst_byt_pcm_open,
 	.close		= sst_byt_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From f70eab3aad004b7496755ec77763bfd6e2eff553 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:28 +0530
Subject: ASoC: nuc900: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/nuc900/nuc900-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index cd0486cad7f5..bd6dfa218d59 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -271,7 +271,7 @@ static int nuc900_dma_mmap(struct snd_pcm_substream *substream,
 			   runtime->dma_addr, runtime->dma_bytes);
 }
 
-static struct snd_pcm_ops nuc900_dma_ops = {
+static const struct snd_pcm_ops nuc900_dma_ops = {
 	.open		= nuc900_dma_open,
 	.close		= nuc900_dma_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From b23bd34c402d08a2c1d6facdba4a75e016601c12 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Mon, 14 Aug 2017 11:26:32 +0530
Subject: ASoC: sh: constify snd_pcm_ops structures

snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/soc.h> work with
const snd_pcm_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/dma-sh7760.c | 2 +-
 sound/soc/sh/fsi.c        | 2 +-
 sound/soc/sh/rcar/core.c  | 2 +-
 sound/soc/sh/siu_pcm.c    | 2 +-
 4 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 81c743322445..35788a6db963 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -294,7 +294,7 @@ static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream)
 	return bytes_to_frames(runtime, pos);
 }
 
-static struct snd_pcm_ops camelot_pcm_ops = {
+static const struct snd_pcm_ops camelot_pcm_ops = {
 	.open		= camelot_pcm_open,
 	.close		= camelot_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 90021b8a799a..5d7d9fe8bc98 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1755,7 +1755,7 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
 	return fsi_sample2frame(fsi, io->buff_sample_pos);
 }
 
-static struct snd_pcm_ops fsi_pcm_ops = {
+static const struct snd_pcm_ops fsi_pcm_ops = {
 	.open		= fsi_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= fsi_hw_params,
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index e50f7e7981c8..08f3a06978e1 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1158,7 +1158,7 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream)
 	return pointer;
 }
 
-static struct snd_pcm_ops rsnd_pcm_ops = {
+static const struct snd_pcm_ops rsnd_pcm_ops = {
 	.open		= rsnd_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= rsnd_hw_params,
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 82902f56e82f..3118cb0ee3f2 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -593,7 +593,7 @@ static void siu_pcm_free(struct snd_pcm *pcm)
 	dev_dbg(pcm->card->dev, "%s\n", __func__);
 }
 
-static struct snd_pcm_ops siu_pcm_ops = {
+static const struct snd_pcm_ops siu_pcm_ops = {
 	.open		= siu_pcm_open,
 	.close		= siu_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-- 
cgit v1.2.3


From 0aef27cab880d9612331350c008aec00cbb3d247 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 12 Aug 2017 13:00:49 +0200
Subject: ASoC: sun4i-i2s: Add clkdiv offsets to quirks

The BCLKDIV and MCLKDIV found on newer SoCs start from an offset of 1.
Add the functionality to adjust the division values according to the
needs to the device being used.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index d7ee7a443e4e..9a35313c4f9b 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -94,9 +94,13 @@
  * struct sun4i_i2s_quirks - Differences between SoC variants.
  *
  * @has_reset: SoC needs reset deasserted.
+ * @mclk_offset: Value by which mclkdiv needs to be adjusted.
+ * @bclk_offset: Value by which bclkdiv needs to be adjusted.
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
+	unsigned int			mclk_offset;
+	unsigned int			bclk_offset;
 };
 
 struct sun4i_i2s {
@@ -237,6 +241,10 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s,
 	if (mclk_div < 0)
 		return -EINVAL;
 
+	/* Adjust the clock division values if needed */
+	bclk_div += i2s->variant->bclk_offset;
+	mclk_div += i2s->variant->mclk_offset;
+
 	regmap_write(i2s->regmap, SUN4I_I2S_CLK_DIV_REG,
 		     SUN4I_I2S_CLK_DIV_BCLK(bclk_div) |
 		     SUN4I_I2S_CLK_DIV_MCLK(mclk_div) |
-- 
cgit v1.2.3


From cd1c63df5fa8196d8ed5138f1ecc22bbabf5f6df Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 12 Aug 2017 13:00:50 +0200
Subject: ASoC: sun4i-i2s: Add regmap config to quirks

The newer SoCs have a larger range than the original SoC that this
driver was developed for. By adding the regmap config to the quirks
then the driver can initialise the managed register map correctly.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 10 +++++++---
 1 file changed, 7 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9a35313c4f9b..f6f3c409f25e 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -94,11 +94,13 @@
  * struct sun4i_i2s_quirks - Differences between SoC variants.
  *
  * @has_reset: SoC needs reset deasserted.
+ * @sun4i_i2s_regmap: regmap config to use.
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
  * @bclk_offset: Value by which bclkdiv needs to be adjusted.
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
+	const struct regmap_config	*sun4i_i2s_regmap;
 	unsigned int			mclk_offset;
 	unsigned int			bclk_offset;
 };
@@ -674,11 +676,13 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
 }
 
 static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
-	.has_reset	= false,
+	.has_reset		= false,
+	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
 };
 
 static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
-	.has_reset	= true,
+	.has_reset		= true,
+	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
 };
 
 static int sun4i_i2s_probe(struct platform_device *pdev)
@@ -717,7 +721,7 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 	}
 
 	i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs,
-					    &sun4i_i2s_regmap_config);
+					    i2s->variant->sun4i_i2s_regmap);
 	if (IS_ERR(i2s->regmap)) {
 		dev_err(&pdev->dev, "Regmap initialisation failed\n");
 		return PTR_ERR(i2s->regmap);
-- 
cgit v1.2.3


From 3509476e522baaa68813609801c48aa428bae81a Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 12 Aug 2017 13:00:51 +0200
Subject: ASoC: sun4i-i2s: Add TX FIFO offset to quirks

It has been seen that the newer SoCs have a different TX FIFO
address. Add this to the quirks structure.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index f6f3c409f25e..dfb794ffff92 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -94,12 +94,14 @@
  * struct sun4i_i2s_quirks - Differences between SoC variants.
  *
  * @has_reset: SoC needs reset deasserted.
+ * @reg_offset_txdata: offset of the tx fifo.
  * @sun4i_i2s_regmap: regmap config to use.
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
  * @bclk_offset: Value by which bclkdiv needs to be adjusted.
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
+	unsigned int			reg_offset_txdata;	/* TX FIFO */
 	const struct regmap_config	*sun4i_i2s_regmap;
 	unsigned int			mclk_offset;
 	unsigned int			bclk_offset;
@@ -677,11 +679,13 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
 
 static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.has_reset		= false,
+	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
 };
 
 static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.has_reset		= true,
+	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
 };
 
@@ -750,7 +754,8 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 		}
 	}
 
-	i2s->playback_dma_data.addr = res->start + SUN4I_I2S_FIFO_TX_REG;
+	i2s->playback_dma_data.addr = res->start +
+					i2s->variant->reg_offset_txdata;
 	i2s->playback_dma_data.maxburst = 8;
 
 	i2s->capture_dma_data.addr = res->start + SUN4I_I2S_FIFO_RX_REG;
-- 
cgit v1.2.3


From 60ea0394a602d5abc24bd2381fa0966a8720eec1 Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 21:45:37 +0200
Subject: ASoC: blackfin: Delete an error message for a failed memory
 allocation in sport_create()

Omit an extra message for a memory allocation failure in this function.

This issue was detected by using the Coccinelle software.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/blackfin/bf6xx-sport.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index dfb744381c42..1bc3e0a47a57 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -388,10 +388,9 @@ struct sport_device *sport_create(struct platform_device *pdev)
 	int ret;
 
 	sport = kzalloc(sizeof(*sport), GFP_KERNEL);
-	if (!sport) {
-		dev_err(dev, "Unable to allocate memory for sport device\n");
+	if (!sport)
 		return NULL;
-	}
+
 	sport->pdev = pdev;
 
 	ret = sport_get_resource(sport);
-- 
cgit v1.2.3


From 11fca34b2783408dbf37e42a38e4b55d54d6fa3a Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 22:02:47 +0200
Subject: ASoC: blackfin: Use common error handling code in sport_create()

Add a jump target so that a bit of exception handling can be better reused
at the end of this function.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/blackfin/bf6xx-sport.c | 15 +++++++--------
 1 file changed, 7 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index 1bc3e0a47a57..fcddebf2558c 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -394,19 +394,18 @@ struct sport_device *sport_create(struct platform_device *pdev)
 	sport->pdev = pdev;
 
 	ret = sport_get_resource(sport);
-	if (ret) {
-		kfree(sport);
-		return NULL;
-	}
+	if (ret)
+		goto free_data;
 
 	ret = sport_request_resource(sport);
-	if (ret) {
-		kfree(sport);
-		return NULL;
-	}
+	if (ret)
+		goto free_data;
 
 	dev_dbg(dev, "SPORT create success\n");
 	return sport;
+free_data:
+	kfree(sport);
+	return NULL;
 }
 EXPORT_SYMBOL(sport_create);
 
-- 
cgit v1.2.3


From 11f25bbaaa87bf583a453b202db9c8ee9b3cbd50 Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 22:20:07 +0200
Subject: ASoC: blackfin: Add some spaces for better code readability

Use space characters at some source code places according to
the Linux coding style convention.

Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/blackfin/bf6xx-sport.c | 18 ++++++++----------
 1 file changed, 8 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index fcddebf2558c..d2caadfe7b6d 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -129,7 +129,7 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
 
 	for (i = 0; i < fragcount; ++i) {
 		desc[i].next_desc_addr  = &(desc[i + 1]);
-		desc[i].start_addr = (unsigned long)buf + i*fragsize;
+		desc[i].start_addr = (unsigned long)buf + i * fragsize;
 		desc[i].cfg = cfg;
 		desc[i].x_count = count;
 		desc[i].x_modify = wdsize;
@@ -138,7 +138,7 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
 	}
 
 	/* make circular */
-	desc[fragcount-1].next_desc_addr = desc;
+	desc[fragcount - 1].next_desc_addr = desc;
 }
 
 int sport_config_tx_dma(struct sport_device *sport, void *buf,
@@ -148,7 +148,7 @@ int sport_config_tx_dma(struct sport_device *sport, void *buf,
 	unsigned int cfg;
 	dma_addr_t addr;
 
-	count = fragsize/sport->wdsize;
+	count = fragsize / sport->wdsize;
 
 	if (sport->tx_desc)
 		dma_free_coherent(NULL, sport->tx_desc_size,
@@ -166,8 +166,7 @@ int sport_config_tx_dma(struct sport_device *sport, void *buf,
 	cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6;
 
 	setup_desc(sport->tx_desc, buf, fragcount, fragsize,
-			cfg|DMAEN, count, sport->wdsize);
-
+		   cfg | DMAEN, count, sport->wdsize);
 	return 0;
 }
 EXPORT_SYMBOL(sport_config_tx_dma);
@@ -179,7 +178,7 @@ int sport_config_rx_dma(struct sport_device *sport, void *buf,
 	unsigned int cfg;
 	dma_addr_t addr;
 
-	count = fragsize/sport->wdsize;
+	count = fragsize / sport->wdsize;
 
 	if (sport->rx_desc)
 		dma_free_coherent(NULL, sport->rx_desc_size,
@@ -198,8 +197,7 @@ int sport_config_rx_dma(struct sport_device *sport, void *buf,
 		| WNR | NDSIZE_6;
 
 	setup_desc(sport->rx_desc, buf, fragcount, fragsize,
-			cfg|DMAEN, count, sport->wdsize);
-
+		   cfg | DMAEN, count, sport->wdsize);
 	return 0;
 }
 EXPORT_SYMBOL(sport_config_rx_dma);
@@ -226,7 +224,7 @@ static irqreturn_t sport_tx_irq(int irq, void *dev_id)
 	static unsigned long status;
 
 	status = get_dma_curr_irqstat(sport->tx_dma_chan);
-	if (status & (DMA_DONE|DMA_ERR)) {
+	if (status & (DMA_DONE | DMA_ERR)) {
 		clear_dma_irqstat(sport->tx_dma_chan);
 		SSYNC();
 	}
@@ -241,7 +239,7 @@ static irqreturn_t sport_rx_irq(int irq, void *dev_id)
 	unsigned long status;
 
 	status = get_dma_curr_irqstat(sport->rx_dma_chan);
-	if (status & (DMA_DONE|DMA_ERR)) {
+	if (status & (DMA_DONE | DMA_ERR)) {
 		clear_dma_irqstat(sport->rx_dma_chan);
 		SSYNC();
 	}
-- 
cgit v1.2.3


From b88845484d05c2910ffc40a7d30fa36e58b2dd0d Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 16:04:53 +0200
Subject: ASoC: dwc: Delete an error message for a failed memory allocation in
 dw_i2s_probe()

Omit an extra message for a memory allocation failure in this function.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/dwc/dwc-i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index eebe9d34e7a0..e27e21f8569a 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -617,10 +617,8 @@ static int dw_i2s_probe(struct platform_device *pdev)
 	const char *clk_id;
 
 	dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
-	if (!dev) {
-		dev_warn(&pdev->dev, "kzalloc fail\n");
+	if (!dev)
 		return -ENOMEM;
-	}
 
 	dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL);
 	if (!dw_i2s_dai)
-- 
cgit v1.2.3


From c7591edc2abd279018abb68aea5bdc604b8f56dd Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 09:35:33 +0200
Subject: ASoC: kirkwood: Delete an error message for a failed memory
 allocation in kirkwood_i2s_dev_probe()

Omit an extra message for a memory allocation failure in this function.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/kirkwood/kirkwood-i2s.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3b09f71b9535..105a73cc5158 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -538,10 +538,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	int err;
 
 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
-	if (!priv) {
-		dev_err(&pdev->dev, "allocation failed\n");
+	if (!priv)
 		return -ENOMEM;
-	}
+
 	dev_set_drvdata(&pdev->dev, priv);
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-- 
cgit v1.2.3


From dd80627d0688709a77233787a9c9d7e1f8e419ac Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Thu, 10 Aug 2017 18:18:20 +0200
Subject: ASoC: fsi: Delete an error message for a failed memory allocation in
 fsi_probe()

Omit an extra message for a memory allocation failure in this function.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/fsi.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 005b21506556..1f41cf1ff830 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1962,10 +1962,8 @@ static int fsi_probe(struct platform_device *pdev)
 	}
 
 	master = devm_kzalloc(&pdev->dev, sizeof(*master), GFP_KERNEL);
-	if (!master) {
-		dev_err(&pdev->dev, "Could not allocate master\n");
+	if (!master)
 		return -ENOMEM;
-	}
 
 	master->base = devm_ioremap_nocache(&pdev->dev,
 					    res->start, resource_size(res));
-- 
cgit v1.2.3


From b48b2710913d583ff93c365413532e1a7cd60d84 Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Thu, 10 Aug 2017 18:38:09 +0200
Subject: ASoC: rockchip: Delete an error message for a failed memory
 allocation in rockchip_i2s_probe()

Omit an extra message for a memory allocation failure in this function.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 199338fdeda0..b6590467fd14 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -579,10 +579,8 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 	int val;
 
 	i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
-	if (!i2s) {
-		dev_err(&pdev->dev, "Can't allocate rk_i2s_dev\n");
+	if (!i2s)
 		return -ENOMEM;
-	}
 
 	i2s->dev = &pdev->dev;
 
-- 
cgit v1.2.3


From dc09233908cf47339f5000dc54574685e917af77 Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 16:51:46 +0200
Subject: ASoC: DaVinci: Delete an error message for a failed memory allocation
 in two functions

Omit an extra message for a memory allocation failure in these functions.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 2 --
 sound/soc/davinci/davinci-vcif.c  | 5 +----
 2 files changed, 1 insertion(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ec1d301ac2..d486908d4e0e 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1602,8 +1602,6 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of(
 		pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata),
 				     GFP_KERNEL);
 		if (!pdata) {
-			dev_err(&pdev->dev,
-				"Failed to allocate memory for pdata\n");
 			ret = -ENOMEM;
 			return pdata;
 		}
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index c77d9218795a..5415b72393fa 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -210,11 +210,8 @@ static int davinci_vcif_probe(struct platform_device *pdev)
 	davinci_vcif_dev = devm_kzalloc(&pdev->dev,
 					sizeof(struct davinci_vcif_dev),
 					GFP_KERNEL);
-	if (!davinci_vcif_dev) {
-		dev_dbg(&pdev->dev,
-			"could not allocate memory for private data\n");
+	if (!davinci_vcif_dev)
 		return -ENOMEM;
-	}
 
 	/* DMA tx params */
 	davinci_vcif_dev->davinci_vc = davinci_vc;
-- 
cgit v1.2.3


From 3013c75fadc581160be8d89c2c0f29219971c58f Mon Sep 17 00:00:00 2001
From: Colin Ian King
Date: Sat, 12 Aug 2017 23:50:28 +0100
Subject: ASoC: hdmi-codec: make a function and two arrays static

The function hdmi_codec_eld_chmap and arrays hdmi_codec_stereo_chmaps
and hdmi_codec_8ch_chmaps are local to the source and do not need to
be in global scope, so make them static.

Cleans up sparse warnings:
symbol 'hdmi_codec_eld_chmap' was not declared. Should it be static?
symbol 'hdmi_codec_stereo_chmaps' was not declared. Should it be static?
symbol 'hdmi_codec_8ch_chmaps' was not declared. Should it be static?

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdmi-codec.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 509ab513b4b2..f7520413c671 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -67,14 +67,14 @@ struct hdmi_codec_cea_spk_alloc {
 };
 
 /* Channel maps  stereo HDMI */
-const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = {
+static const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = {
 	{ .channels = 2,
 	  .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } },
 	{ }
 };
 
 /* Channel maps for multi-channel playbacks, up to 8 n_ch */
-const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = {
+static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = {
 	{ .channels = 2, /* CA_ID 0x00 */
 	  .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } },
 	{ .channels = 4, /* CA_ID 0x01 */
@@ -340,7 +340,7 @@ static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc)
 	return spk_mask;
 }
 
-void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp)
+static void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp)
 {
 	u8 spk_alloc;
 	unsigned long spk_mask;
-- 
cgit v1.2.3


From 443be77e4b09d98c28196dca7132a1191ecf742b Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 15:10:43 +0200
Subject: ASoC: Freescale: Delete an error message for a failed memory
 allocation in three functions

Omit an extra message for a memory allocation failure in these functions.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_asrc_dma.c | 4 +---
 sound/soc/fsl/fsl_dma.c      | 1 -
 sound/soc/fsl/fsl_ssi.c      | 4 +---
 3 files changed, 2 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 282d841840b1..2baf19608bd0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -279,10 +279,8 @@ static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream)
 	struct fsl_asrc_pair *pair;
 
 	pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL);
-	if (!pair) {
-		dev_err(dev, "failed to allocate pair\n");
+	if (!pair)
 		return -ENOMEM;
-	}
 
 	pair->asrc_priv = asrc_priv;
 
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 979ed1b3ecc4..c17359648915 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -905,7 +905,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev)
 
 	dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
 	if (!dma) {
-		dev_err(&pdev->dev, "could not allocate dma object\n");
 		of_node_put(ssi_np);
 		return -ENOMEM;
 	}
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 173cb8496641..64598d1183f8 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1432,10 +1432,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 
 	ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private),
 			GFP_KERNEL);
-	if (!ssi_private) {
-		dev_err(&pdev->dev, "could not allocate DAI object\n");
+	if (!ssi_private)
 		return -ENOMEM;
-	}
 
 	ssi_private->soc = of_id->data;
 	ssi_private->dev = &pdev->dev;
-- 
cgit v1.2.3


From a4fe6f37269623b3be21e3697d59677809ef512b Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Tue, 15 Aug 2017 17:10:08 +0200
Subject: ASoC: blackfin: constify snd_soc_dai_ops structures

These snd_soc_dai_ops structures are only stored in the ops field of
a snd_soc_dai_driver structure, which is const.  Thus, the
snd_soc_dai_ops structures can be const too.

Done with the help of Coccinelle.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/blackfin/bf6xx-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c
index bd3b4d464145..819cff149a25 100644
--- a/sound/soc/blackfin/bf6xx-i2s.c
+++ b/sound/soc/blackfin/bf6xx-i2s.c
@@ -164,7 +164,7 @@ static int bfin_i2s_resume(struct snd_soc_dai *dai)
 #define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops bfin_i2s_dai_ops = {
+static const struct snd_soc_dai_ops bfin_i2s_dai_ops = {
 	.hw_params	= bfin_i2s_hw_params,
 	.set_fmt	= bfin_i2s_set_dai_fmt,
 };
-- 
cgit v1.2.3


From 0966a2588779d0d44706f973c1e8ff240be43017 Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Tue, 15 Aug 2017 17:10:09 +0200
Subject: ASoC: rockchip: constify snd_soc_dai_ops structures

These snd_soc_dai_ops structures are only stored in the ops field of
a snd_soc_dai_driver structure, which is const.  Thus, the
snd_soc_dai_ops structures can be const too.

Done with the help of Coccinelle.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_pdm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index c5ddeed97260..400e29edb1c9 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -249,7 +249,7 @@ static int rockchip_pdm_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops rockchip_pdm_dai_ops = {
+static const struct snd_soc_dai_ops rockchip_pdm_dai_ops = {
 	.set_fmt = rockchip_pdm_set_fmt,
 	.trigger = rockchip_pdm_trigger,
 	.hw_params = rockchip_pdm_hw_params,
-- 
cgit v1.2.3


From a6a67da3da332dbde7c04cba9fccef04648c704a Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Tue, 15 Aug 2017 17:10:10 +0200
Subject: ASoC: codecs: es8316: constify snd_soc_dai_ops structures

These snd_soc_dai_ops structures are only stored in the ops field of
a snd_soc_dai_driver structure, which is const.  Thus, the
snd_soc_dai_ops structures can be const too.

Done with the help of Coccinelle.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8316.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index ecc02449c569..f5cf48c9c3b0 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -502,7 +502,7 @@ static int es8316_mute(struct snd_soc_dai *dai, int mute)
 #define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops es8316_ops = {
+static const struct snd_soc_dai_ops es8316_ops = {
 	.startup = es8316_pcm_startup,
 	.hw_params = es8316_pcm_hw_params,
 	.set_fmt = es8316_set_dai_fmt,
-- 
cgit v1.2.3


From 66f41da341cd97ad362c4a40e1c9a0a28d746f81 Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Tue, 15 Aug 2017 17:10:11 +0200
Subject: ASoC: codecs: zx_aud96p22: constify snd_soc_dai_ops structures

These snd_soc_dai_ops structures are only stored in the ops field of
a snd_soc_dai_driver structure, which is const.  Thus, the
snd_soc_dai_ops structures can be const too.

Done with the help of Coccinelle.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/zx_aud96p22.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c
index bc23ce1a30a1..0e5b43b948b4 100644
--- a/sound/soc/codecs/zx_aud96p22.c
+++ b/sound/soc/codecs/zx_aud96p22.c
@@ -312,7 +312,7 @@ static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static struct snd_soc_dai_ops aud96p22_dai_ops = {
+static const struct snd_soc_dai_ops aud96p22_dai_ops = {
 	.set_fmt = aud96p22_set_fmt,
 };
 
-- 
cgit v1.2.3


From a1150d40b219af1344f7bcc9b365dc5ec7bfbb65 Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Tue, 15 Aug 2017 17:10:12 +0200
Subject: ASoC: spear: constify snd_soc_dai_ops structures

These snd_soc_dai_ops structures are only stored in the ops field of
a snd_soc_dai_driver structure, which is const.  Thus, the
snd_soc_dai_ops structures can be const too.

Done with the help of Coccinelle.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/spear/spdif_in.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 01b28e59b4c0..78a6a360b4a6 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -151,7 +151,7 @@ static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
-static struct snd_soc_dai_ops spdif_in_dai_ops = {
+static const struct snd_soc_dai_ops spdif_in_dai_ops = {
 	.shutdown	= spdif_in_shutdown,
 	.trigger	= spdif_in_trigger,
 	.hw_params	= spdif_in_hw_params,
-- 
cgit v1.2.3


From 8937ea0f3c3a90da3c594a4e2b038d2c7b282eec Mon Sep 17 00:00:00 2001
From: Lori Hikichi
Date: Mon, 14 Aug 2017 15:06:49 -0700
Subject: ASoC: cygnus: Add support for 384kHz frame rates

Allow the audio ports to operate at 384kHz.

Signed-off-by: Lori Hikichi <lori.hikichi@broadcom.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/bcm/cygnus-ssp.c | 235 +++++++++++----------------------------------
 1 file changed, 55 insertions(+), 180 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index e710bb0c5637..1a57a4ebc8f0 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -27,12 +27,6 @@
 
 #define DEFAULT_VCO    1354750204
 
-#define CYGNUS_TDM_RATE \
-		(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
-		SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_22050 | \
-		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-		SNDRV_PCM_RATE_48000)
-
 #define CAPTURE_FCI_ID_BASE 0x180
 #define CYGNUS_SSP_TRISTATE_MASK 0x001fff
 #define CYGNUS_PLLCLKSEL_MASK 0xf
@@ -234,152 +228,20 @@ static const struct pll_macro_entry pll_predef_mclk[] = {
 	{98304000, 2},
 };
 
+#define CYGNUS_RATE_MIN     8000
+#define CYGNUS_RATE_MAX   384000
+
 /* List of valid frame sizes for tdm mode */
 static const int ssp_valid_tdm_framesize[] = {32, 64, 128, 256, 512};
 
-/*
- * Use this relationship to derive the sampling rate (lrclk)
- * lrclk = (mclk) / ((2*mclk_to_sclk_ratio) * (32 * SCLK))).
- *
- * Use mclk and pll_ch from the table above
- *
- * Valid SCLK = 0/1/2/4/8/12
- *
- * mclk_to_sclk_ratio = number of MCLK per SCLK. Division is twice the
- * value programmed in this field.
- * Valid mclk_to_sclk_ratio = 1 through to 15
- *
- * eg: To set lrclk = 48khz, set mclk = 12288000, mclk_to_sclk_ratio = 2,
- * SCLK = 64
- */
-struct _ssp_clk_coeff {
-	u32 mclk;
-	u32 sclk_rate;
-	u32 rate;
-	u32 mclk_rate;
+static const unsigned int cygnus_rates[] = {
+	 8000, 11025,  16000,  22050,  32000,  44100, 48000,
+	88200, 96000, 176400, 192000, 352800, 384000
 };
 
-static const struct _ssp_clk_coeff ssp_clk_coeff[] = {
-	{ 4096000,  32,  16000, 4},
-	{ 4096000,  32,  32000, 2},
-	{ 4096000,  64,   8000, 4},
-	{ 4096000,  64,  16000, 2},
-	{ 4096000,  64,  32000, 1},
-	{ 4096000, 128,   8000, 2},
-	{ 4096000, 128,  16000, 1},
-	{ 4096000, 256,   8000, 1},
-
-	{ 6144000,  32,  16000, 6},
-	{ 6144000,  32,  32000, 3},
-	{ 6144000,  32,  48000, 2},
-	{ 6144000,  32,  96000, 1},
-	{ 6144000,  64,   8000, 6},
-	{ 6144000,  64,  16000, 3},
-	{ 6144000,  64,  48000, 1},
-	{ 6144000, 128,   8000, 3},
-
-	{ 8192000,  32,  32000, 4},
-	{ 8192000,  64,  16000, 4},
-	{ 8192000,  64,  32000, 2},
-	{ 8192000, 128,   8000, 4},
-	{ 8192000, 128,  16000, 2},
-	{ 8192000, 128,  32000, 1},
-	{ 8192000, 256,   8000, 2},
-	{ 8192000, 256,  16000, 1},
-	{ 8192000, 512,   8000, 1},
-
-	{12288000,  32,  32000, 6},
-	{12288000,  32,  48000, 4},
-	{12288000,  32,  96000, 2},
-	{12288000,  32, 192000, 1},
-	{12288000,  64,  16000, 6},
-	{12288000,  64,  32000, 3},
-	{12288000,  64,  48000, 2},
-	{12288000,  64,  96000, 1},
-	{12288000, 128,   8000, 6},
-	{12288000, 128,  16000, 3},
-	{12288000, 128,  48000, 1},
-	{12288000, 256,   8000, 3},
-
-	{16384000,  64,  32000, 4},
-	{16384000, 128,  16000, 4},
-	{16384000, 128,  32000, 2},
-	{16384000, 256,   8000, 4},
-	{16384000, 256,  16000, 2},
-	{16384000, 256,  32000, 1},
-	{16384000, 512,   8000, 2},
-	{16384000, 512,  16000, 1},
-
-	{24576000,  32,  96000, 4},
-	{24576000,  32, 192000, 2},
-	{24576000,  64,  32000, 6},
-	{24576000,  64,  48000, 4},
-	{24576000,  64,  96000, 2},
-	{24576000,  64, 192000, 1},
-	{24576000, 128,  16000, 6},
-	{24576000, 128,  32000, 3},
-	{24576000, 128,  48000, 2},
-	{24576000, 256,   8000, 6},
-	{24576000, 256,  16000, 3},
-	{24576000, 256,  48000, 1},
-	{24576000, 512,   8000, 3},
-
-	{49152000,  32, 192000, 4},
-	{49152000,  64,  96000, 4},
-	{49152000,  64, 192000, 2},
-	{49152000, 128,  32000, 6},
-	{49152000, 128,  48000, 4},
-	{49152000, 128,  96000, 2},
-	{49152000, 128, 192000, 1},
-	{49152000, 256,  16000, 6},
-	{49152000, 256,  32000, 3},
-	{49152000, 256,  48000, 2},
-	{49152000, 256,  96000, 1},
-	{49152000, 512,   8000, 6},
-	{49152000, 512,  16000, 3},
-	{49152000, 512,  48000, 1},
-
-	{ 5644800,  32,  22050, 4},
-	{ 5644800,  32,  44100, 2},
-	{ 5644800,  32,  88200, 1},
-	{ 5644800,  64,  11025, 4},
-	{ 5644800,  64,  22050, 2},
-	{ 5644800,  64,  44100, 1},
-
-	{11289600,  32,  44100, 4},
-	{11289600,  32,  88200, 2},
-	{11289600,  32, 176400, 1},
-	{11289600,  64,  22050, 4},
-	{11289600,  64,  44100, 2},
-	{11289600,  64,  88200, 1},
-	{11289600, 128,  11025, 4},
-	{11289600, 128,  22050, 2},
-	{11289600, 128,  44100, 1},
-
-	{22579200,  32,  88200, 4},
-	{22579200,  32, 176400, 2},
-	{22579200,  64,  44100, 4},
-	{22579200,  64,  88200, 2},
-	{22579200,  64, 176400, 1},
-	{22579200, 128,  22050, 4},
-	{22579200, 128,  44100, 2},
-	{22579200, 128,  88200, 1},
-	{22579200, 256,  11025, 4},
-	{22579200, 256,  22050, 2},
-	{22579200, 256,  44100, 1},
-
-	{45158400,  32, 176400, 4},
-	{45158400,  64,  88200, 4},
-	{45158400,  64, 176400, 2},
-	{45158400, 128,  44100, 4},
-	{45158400, 128,  88200, 2},
-	{45158400, 128, 176400, 1},
-	{45158400, 256,  22050, 4},
-	{45158400, 256,  44100, 2},
-	{45158400, 256,  88200, 1},
-	{45158400, 512,  11025, 4},
-	{45158400, 512,  22050, 2},
-	{45158400, 512,  44100, 1},
+static const struct snd_pcm_hw_constraint_list cygnus_rate_constraint = {
+	.count = ARRAY_SIZE(cygnus_rates),
+	.list = cygnus_rates,
 };
 
 static struct cygnus_aio_port *cygnus_dai_get_portinfo(struct snd_soc_dai *dai)
@@ -679,40 +541,55 @@ static int pll_configure_mclk(struct cygnus_audio *cygaud, u32 mclk,
 	return p_entry->pll_ch_num;
 }
 
-static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio,
-			struct cygnus_audio *cygaud)
+static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio)
 {
-	u32 value, i = 0;
+	u32 value;
 	u32 mask = 0xf;
 	u32 sclk;
-	bool found = false;
-	const struct _ssp_clk_coeff *p_entry = NULL;
+	u32 mclk_rate;
+	unsigned int bit_rate;
+	unsigned int ratio;
 
-	for (i = 0; i < ARRAY_SIZE(ssp_clk_coeff); i++) {
-		p_entry = &ssp_clk_coeff[i];
-		if ((p_entry->rate == aio->lrclk) &&
-		    (p_entry->sclk_rate == aio->bit_per_frame) &&
-		    (p_entry->mclk == aio->mclk)) {
-			found = true;
-			break;
-		}
-	}
-	if (!found) {
+	bit_rate = aio->bit_per_frame * aio->lrclk;
+
+	/*
+	 * Check if the bit clock can be generated from the given MCLK.
+	 * MCLK must be a perfect multiple of bit clock and must be one of the
+	 * following values... (2,4,6,8,10,12,14)
+	 */
+	if ((aio->mclk % bit_rate) != 0)
+		return -EINVAL;
+
+	ratio = aio->mclk / bit_rate;
+	switch (ratio) {
+	case 2:
+	case 4:
+	case 6:
+	case 8:
+	case 10:
+	case 12:
+	case 14:
+		mclk_rate = ratio / 2;
+		break;
+
+	default:
 		dev_err(aio->cygaud->dev,
-			"No valid match found in ssp_clk_coeff array\n");
+			"Invalid combination of MCLK and BCLK\n");
 		dev_err(aio->cygaud->dev, "lrclk = %u, bits/frame = %u, mclk = %u\n",
 			aio->lrclk, aio->bit_per_frame, aio->mclk);
 		return -EINVAL;
 	}
 
-	sclk = aio->bit_per_frame;
-	if (sclk == 512)
-		sclk = 0;
-	/* sclks_per_1fs_div = sclk cycles/32 */
-	sclk /= 32;
 	/* Set sclk rate */
 	switch (aio->port_type) {
 	case PORT_TDM:
+		sclk = aio->bit_per_frame;
+		if (sclk == 512)
+			sclk = 0;
+
+		/* sclks_per_1fs_div = sclk cycles/32 */
+		sclk /= 32;
+
 		/* Set number of bitclks per frame */
 		value = readl(aio->cygaud->audio + aio->regs.i2s_cfg);
 		value &= ~(mask << I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32);
@@ -731,7 +608,7 @@ static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio,
 	/* Set MCLK_RATE ssp port (spdif and ssp are the same) */
 	value = readl(aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
 	value &= ~(0xf << I2S_OUT_MCLKRATE_SHIFT);
-	value |= (p_entry->mclk_rate << I2S_OUT_MCLKRATE_SHIFT);
+	value |= (mclk_rate << I2S_OUT_MCLKRATE_SHIFT);
 	writel(value, aio->cygaud->audio + aio->regs.i2s_mclk_cfg);
 
 	dev_dbg(aio->cygaud->dev, "mclk cfg reg = 0x%x\n", value);
@@ -745,7 +622,6 @@ static int cygnus_ssp_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
-	struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai);
 	int rate, bitres;
 	u32 value;
 	u32 mask = 0x1f;
@@ -841,7 +717,7 @@ static int cygnus_ssp_hw_params(struct snd_pcm_substream *substream,
 	aio->lrclk = rate;
 
 	if (!aio->is_slave)
-		ret = cygnus_ssp_set_clocks(aio, cygaud);
+		ret = cygnus_ssp_set_clocks(aio);
 
 	return ret;
 }
@@ -888,6 +764,11 @@ static int cygnus_ssp_startup(struct snd_pcm_substream *substream,
 	else
 		aio->clk_trace.cap_en = true;
 
+	substream->runtime->hw.rate_min = CYGNUS_RATE_MIN;
+	substream->runtime->hw.rate_max = CYGNUS_RATE_MAX;
+
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &cygnus_rate_constraint);
 	return 0;
 }
 
@@ -1261,9 +1142,7 @@ static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = {
 	.playback = { \
 		.channels_min = 1, \
 		.channels_max = 16, \
-		.rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 | \
-			SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
-			SNDRV_PCM_RATE_192000, \
+		.rates = SNDRV_PCM_RATE_KNOT, \
 		.formats = SNDRV_PCM_FMTBIT_S8 | \
 				SNDRV_PCM_FMTBIT_S16_LE | \
 				SNDRV_PCM_FMTBIT_S32_LE, \
@@ -1271,9 +1150,7 @@ static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = {
 	.capture = { \
 		.channels_min = 2, \
 		.channels_max = 16, \
-		.rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 | \
-			SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
-			SNDRV_PCM_RATE_192000, \
+		.rates = SNDRV_PCM_RATE_KNOT, \
 		.formats =  SNDRV_PCM_FMTBIT_S16_LE | \
 					SNDRV_PCM_FMTBIT_S32_LE, \
 	}, \
@@ -1293,9 +1170,7 @@ static struct snd_soc_dai_driver cygnus_spdif_dai_info = {
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 |
-			SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
-			SNDRV_PCM_RATE_192000,
+		.rates = SNDRV_PCM_RATE_KNOT,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE |
 			SNDRV_PCM_FMTBIT_S32_LE,
 	},
-- 
cgit v1.2.3


From 7b814a7d4e83b0917efef9dd11a8c095371f987c Mon Sep 17 00:00:00 2001
From: Anton Vasilyev
Date: Tue, 15 Aug 2017 15:19:54 +0300
Subject: ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove

If (quirks & QUIRK_SEC_DAI == 0) then samsung_i2s_probe() doesn't allocate
sec_dai and pri_dai->sec_dai remains Null, but samsung_i2s_remove()
performs pri_dai->sec_dai dereference in any case.

The patch removes useless reinitialization of sec_dai at
samsung_i2s_remove(), because resources are under devm control.

Found by Linux Driver Verification project (linuxtesting.org).

Signed-off-by: Anton Vasilyev <vasilyev@ispras.ru>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/i2s.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index daf7b892c967..10a4da06c0a1 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -1388,13 +1388,9 @@ err_disable_clk:
 
 static int samsung_i2s_remove(struct platform_device *pdev)
 {
-	struct i2s_dai *pri_dai, *sec_dai;
+	struct i2s_dai *pri_dai;
 
 	pri_dai = dev_get_drvdata(&pdev->dev);
-	sec_dai = pri_dai->sec_dai;
-
-	pri_dai->sec_dai = NULL;
-	sec_dai->pri_dai = NULL;
 
 	pm_runtime_get_sync(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
-- 
cgit v1.2.3


From d0148eb4a41e65ac95bfaaa13c75364b95b1e052 Mon Sep 17 00:00:00 2001
From: Daniel Baluta
Date: Fri, 11 Aug 2017 17:12:19 +0300
Subject: ASoC: simple-scu-card: Parse off codec widgets

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/simple-scu-card.txt | 1 +
 sound/soc/generic/simple-scu-card.c                         | 4 ++++
 2 files changed, 5 insertions(+)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt
index 327d229a51b2..32f8dbce5241 100644
--- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-scu-card.txt
@@ -24,6 +24,7 @@ Optional subnode properties:
 - simple-audio-card,convert-rate	: platform specified sampling rate convert
 - simple-audio-card,convert-channels	: platform specified converted channel size (2 - 8 ch)
 - simple-audio-card,prefix		: see routing
+- simple-audio-card,widgets		: Please refer to widgets.txt.
 - simple-audio-card,routing		: A list of the connections between audio components.
 					  Each entry is a pair of strings, the first being the connection's sink,
 					  the second being the connection's source. Valid names for sources.
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index a75b385455c4..0bb3da755532 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -191,6 +191,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv)
 	if (!node)
 		return -EINVAL;
 
+	ret = asoc_simple_card_of_parse_widgets(card, PREFIX);
+	if (ret < 0)
+		return ret;
+
 	ret = asoc_simple_card_of_parse_routing(card, PREFIX, 0);
 	if (ret < 0)
 		return ret;
-- 
cgit v1.2.3


From 2d5f8487818b9231dfb244adda5e890e004b4a76 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Fri, 11 Aug 2017 21:26:49 +0530
Subject: ASoC: Intel: kbl: make snd_pcm_hw_constraint_list const

Make these const as they are only passed to the function
snd_pcm_hw_constraint_list having the corresponding argument as const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 12b34b9a0104..8bdd6131c6ad 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -286,7 +286,7 @@ static unsigned int rates[] = {
 	48000,
 };
 
-static struct snd_pcm_hw_constraint_list constraints_rates = {
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
 	.count = ARRAY_SIZE(rates),
 	.list  = rates,
 	.mask = 0,
@@ -296,7 +296,7 @@ static unsigned int channels[] = {
 	2,
 };
 
-static struct snd_pcm_hw_constraint_list constraints_channels = {
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
 	.count = ARRAY_SIZE(channels),
 	.list = channels,
 	.mask = 0,
@@ -425,7 +425,7 @@ static unsigned int rates_16000[] = {
 	16000,
 };
 
-static struct snd_pcm_hw_constraint_list constraints_16000 = {
+static const struct snd_pcm_hw_constraint_list constraints_16000 = {
 	.count = ARRAY_SIZE(rates_16000),
 	.list  = rates_16000,
 };
-- 
cgit v1.2.3


From 2788808a1bb4dd0b20c4cea2a98b9fda4c354682 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Fri, 11 Aug 2017 21:57:15 +0530
Subject: ASoC: Intel: Skylake: make skl_dsp_fw_ops const

Make these const as they are only used during a copy operation.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/bxt-sst.c | 2 +-
 sound/soc/intel/skylake/cnl-sst.c | 2 +-
 sound/soc/intel/skylake/skl-sst.c | 4 ++--
 3 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index 71042aceb25f..5e1ac99ffa8b 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -530,7 +530,7 @@ static int bxt_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id)
 	return 0;
 }
 
-static struct skl_dsp_fw_ops bxt_fw_ops = {
+static const struct skl_dsp_fw_ops bxt_fw_ops = {
 	.set_state_D0 = bxt_set_dsp_D0,
 	.set_state_D3 = bxt_set_dsp_D3,
 	.set_state_D0i3 = bxt_schedule_dsp_D0i3,
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
index 7dcfe66a20e2..48ef04d5da89 100644
--- a/sound/soc/intel/skylake/cnl-sst.c
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -277,7 +277,7 @@ static unsigned int cnl_get_errno(struct sst_dsp *ctx)
 	return sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE);
 }
 
-static struct skl_dsp_fw_ops cnl_fw_ops = {
+static const struct skl_dsp_fw_ops cnl_fw_ops = {
 	.set_state_D0 = cnl_set_dsp_D0,
 	.set_state_D3 = cnl_set_dsp_D3,
 	.load_fw = cnl_load_base_firmware,
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 6cc586d3f03b..a7ad734f591d 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -503,7 +503,7 @@ static void skl_clear_module_table(struct sst_dsp *ctx)
 	}
 }
 
-static struct skl_dsp_fw_ops skl_fw_ops = {
+static const struct skl_dsp_fw_ops skl_fw_ops = {
 	.set_state_D0 = skl_set_dsp_D0,
 	.set_state_D3 = skl_set_dsp_D3,
 	.load_fw = skl_load_base_firmware,
@@ -512,7 +512,7 @@ static struct skl_dsp_fw_ops skl_fw_ops = {
 	.unload_mod = skl_unload_module,
 };
 
-static struct skl_dsp_fw_ops kbl_fw_ops = {
+static const struct skl_dsp_fw_ops kbl_fw_ops = {
 	.set_state_D0 = skl_set_dsp_D0,
 	.set_state_D3 = skl_set_dsp_D3,
 	.load_fw = skl_load_base_firmware,
-- 
cgit v1.2.3


From 70bad123ea5bec47329bc8681cbb8e8431188ba2 Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Sat, 12 Aug 2017 08:33:02 +0200
Subject: ASoC: Intel: constify snd_compr_codec_caps structures

These snd_compr_codec_caps structures are only copied into other
structures, so they can be const.

Done with the help of Coccinelle.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/atom/sst/sst_drv_interface.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c
index ce689c5af5ab..71af5449be90 100644
--- a/sound/soc/intel/atom/sst/sst_drv_interface.c
+++ b/sound/soc/intel/atom/sst/sst_drv_interface.c
@@ -407,7 +407,7 @@ static int sst_cdev_caps(struct snd_compr_caps *caps)
 	return 0;
 }
 
-static struct snd_compr_codec_caps caps_mp3 = {
+static const struct snd_compr_codec_caps caps_mp3 = {
 	.num_descriptors = 1,
 	.descriptor[0].max_ch = 2,
 	.descriptor[0].sample_rates[0] = 48000,
@@ -424,7 +424,7 @@ static struct snd_compr_codec_caps caps_mp3 = {
 	.descriptor[0].formats = 0,
 };
 
-static struct snd_compr_codec_caps caps_aac = {
+static const struct snd_compr_codec_caps caps_aac = {
 	.num_descriptors = 2,
 	.descriptor[1].max_ch = 2,
 	.descriptor[0].sample_rates[0] = 48000,
-- 
cgit v1.2.3


From 92ec46da6e35f5f7ff39741517a9466c82d1afb9 Mon Sep 17 00:00:00 2001
From: Markus Elfring
Date: Fri, 11 Aug 2017 11:25:41 +0200
Subject: ASoC: Medfield: Delete an error message for a failed memory
 allocation in snd_mfld_mc_probe()

Omit an extra message for a memory allocation failure in this function.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/mfld_machine.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c
index 4e08885f37aa..6f44acfb4aae 100644
--- a/sound/soc/intel/boards/mfld_machine.c
+++ b/sound/soc/intel/boards/mfld_machine.c
@@ -376,10 +376,8 @@ static int snd_mfld_mc_probe(struct platform_device *pdev)
 	/* audio interrupt base of SRAM location where
 	 * interrupts are stored by System FW */
 	mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
-	if (!mc_drv_ctx) {
-		pr_err("allocation failed\n");
+	if (!mc_drv_ctx)
 		return -ENOMEM;
-	}
 
 	irq_mem = platform_get_resource_byname(
 				pdev, IORESOURCE_MEM, "IRQ_BASE");
-- 
cgit v1.2.3


From 4dbcc5ff2513f2192dc2a5dfdbb9ad3d7e891e85 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 16:09:01 +0530
Subject: ASoC: codecs: make snd_compr_ops const

Make these const as they are only stored in the compr_ops field of a
snd_soc_platform_driver structure, which is of type const.
Done using Coccinelle

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs47l24.c | 2 +-
 sound/soc/codecs/wm5102.c  | 2 +-
 sound/soc/codecs/wm5110.c  | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 47e6fddef92b..c1cbe23169d7 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -1203,7 +1203,7 @@ static struct snd_soc_codec_driver soc_codec_dev_cs47l24 = {
 	},
 };
 
-static struct snd_compr_ops cs47l24_compr_ops = {
+static const struct snd_compr_ops cs47l24_compr_ops = {
 	.open = cs47l24_open,
 	.free = wm_adsp_compr_free,
 	.set_params = wm_adsp_compr_set_params,
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 1fe358e6be61..a8bdbe464f2e 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -2017,7 +2017,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm5102 = {
 	},
 };
 
-static struct snd_compr_ops wm5102_compr_ops = {
+static const struct snd_compr_ops wm5102_compr_ops = {
 	.open = wm5102_open,
 	.free = wm_adsp_compr_free,
 	.set_params = wm_adsp_compr_set_params,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 1bc942152eff..8aa136c28ec1 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2372,7 +2372,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm5110 = {
 	},
 };
 
-static struct snd_compr_ops wm5110_compr_ops = {
+static const struct snd_compr_ops wm5110_compr_ops = {
 	.open = wm5110_open,
 	.free = wm_adsp_compr_free,
 	.set_params = wm_adsp_compr_set_params,
-- 
cgit v1.2.3


From 2eccea8cdeb8fe455fc3a45b4e097f118449c3ef Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Mon, 7 Aug 2017 02:06:55 +0000
Subject: ASoC: soc-core: snd_soc_unregister_component() unregister all
 component

Current snd_soc_unregister_component() unregisters first found
component only which was specified by dev.
This style can't specify concrete component if system registered some
component with same dev. And system need to call this function many
times.
This patch unregister all related component by 1 call.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 12 ++++++++++--
 1 file changed, 10 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b1860b14b075..50c8dba54649 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3365,11 +3365,12 @@ err_free:
 EXPORT_SYMBOL_GPL(snd_soc_register_component);
 
 /**
- * snd_soc_unregister_component - Unregister a component from the ASoC core
+ * snd_soc_unregister_component - Unregister all related component
+ * from the ASoC core
  *
  * @dev: The device to unregister
  */
-void snd_soc_unregister_component(struct device *dev)
+static int __snd_soc_unregister_component(struct device *dev)
 {
 	struct snd_soc_component *component;
 	int found = 0;
@@ -3391,6 +3392,13 @@ void snd_soc_unregister_component(struct device *dev)
 		snd_soc_component_cleanup(component);
 		kfree(component);
 	}
+
+	return found;
+}
+
+void snd_soc_unregister_component(struct device *dev)
+{
+	while (__snd_soc_unregister_component(dev));
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
 
-- 
cgit v1.2.3


From aeb6fa0f15c71a18f757d1686822b6c34cf81399 Mon Sep 17 00:00:00 2001
From: Peng Donglin
Date: Wed, 16 Aug 2017 22:47:53 +0800
Subject: ASoC: compress: Set reasonable compress id string

For dynamic compress rtd, the codec_dai may not have any actual
meaning(like snd-soc-dummy-dai), so compress id can just use the
value of dai_link->stream_name. But for others, its codec_dai
has actual meaning, so compress id can include codec_dai name.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-compress.c | 16 ++++++----------
 1 file changed, 6 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 6c867d50e036..2cb8d3b55fbc 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 	}
 
 	/* check client and interface hw capabilities */
-	snprintf(new_name, sizeof(new_name), "%s %s-%d",
-			rtd->dai_link->stream_name, codec_dai->name, num);
-
 	if (codec_dai->driver->playback.channels_min)
 		playback = 1;
 	if (codec_dai->driver->capture.channels_min)
@@ -758,7 +755,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 		return -EINVAL;
 	}
 
-	if(playback)
+	if (playback)
 		direction = SND_COMPRESS_PLAYBACK;
 	else
 		direction = SND_COMPRESS_CAPTURE;
@@ -794,19 +791,18 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 		else if (rtd->dai_link->dpcm_capture)
 			be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
 		memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops));
-	} else
+	} else {
+		snprintf(new_name, sizeof(new_name), "%s %s-%d",
+			rtd->dai_link->stream_name, codec_dai->name, num);
+
 		memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops));
+	}
 
 	/* Add copy callback for not memory mapped DSPs */
 	if (platform->driver->compr_ops && platform->driver->compr_ops->copy)
 		compr->ops->copy = soc_compr_copy;
 
 	mutex_init(&compr->lock);
-
-	snprintf(new_name, sizeof(new_name), "%s %s-%d",
-		 rtd->dai_link->stream_name,
-		 rtd->codec_dai->name, num);
-
 	ret = snd_compress_new(rtd->card->snd_card, num, direction,
 				new_name, compr);
 	if (ret < 0) {
-- 
cgit v1.2.3


From 393a829bb2f7e12403cb7ff3b62539c27a543a2c Mon Sep 17 00:00:00 2001
From: Harsha Priya N
Date: Wed, 16 Aug 2017 15:37:11 -0700
Subject: ASoC: Intel: kbl_rt5663_rt5514_max98927: Add rt5514 spi dailink

This patch adds a dai to rt5514-spi driver for wake on voice functionality.

Signed-off-by: Hsin-yu Chao <hychao@google.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/Kconfig                             | 3 ++-
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 9 +++++++++
 2 files changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index b301bfff1c09..b3c7f554ec30 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -255,11 +255,12 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH
 
 config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH
         tristate "ASoC Audio driver for KBL with RT5663, RT5514 and MAX98927 in I2S Mode"
-        depends on X86_INTEL_LPSS && I2C
+        depends on X86_INTEL_LPSS && I2C && SPI
         select SND_SOC_INTEL_SST
         select SND_SOC_INTEL_SKYLAKE
         select SND_SOC_RT5663
         select SND_SOC_RT5514
+        select SND_SOC_RT5514_SPI
         select SND_SOC_MAX98927
         select SND_SOC_HDAC_HDMI
         help
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 5ec942885b69..b51dff6fe25b 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -65,6 +65,7 @@ enum {
 	KBL_DPCM_AUDIO_HS_PB,
 	KBL_DPCM_AUDIO_ECHO_REF_CP,
 	KBL_DPCM_AUDIO_DMIC_CP,
+	KBL_DPCM_AUDIO_RT5514_DSP,
 	KBL_DPCM_AUDIO_HDMI1_PB,
 	KBL_DPCM_AUDIO_HDMI2_PB,
 };
@@ -480,6 +481,14 @@ static struct snd_soc_dai_link kabylake_dais[] = {
 		.capture_only = 1,
 		.nonatomic = 1,
 	},
+	[KBL_DPCM_AUDIO_RT5514_DSP] = {
+		.name = "rt5514 dsp",
+		.stream_name = "Wake on Voice",
+		.cpu_dai_name = "spi-PRP0001:00",
+		.platform_name = "spi-PRP0001:00",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+	},
 	[KBL_DPCM_AUDIO_DMIC_CP] = {
 		.name = "Kbl Audio DMIC cap",
 		.stream_name = "dmiccap",
-- 
cgit v1.2.3


From f77bb3b778e85d3477a9e3e0236545663f16793e Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 17 Aug 2017 15:46:07 +0530
Subject: ASoC: fsl: make snd_pcm_hardware const

Make these const as they are only passed as the 2nd argument to the
function snd_soc_set_runtime_hwparams, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_asrc_dma.c | 2 +-
 sound/soc/fsl/imx-pcm-fiq.c  | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 2baf19608bd0..e1b97e59275a 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -20,7 +20,7 @@
 
 #define FSL_ASRC_DMABUF_SIZE	(256 * 1024)
 
-static struct snd_pcm_hardware snd_imx_hardware = {
+static const struct snd_pcm_hardware snd_imx_hardware = {
 	.info = SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		SNDRV_PCM_INFO_MMAP |
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index aef1f7819c40..4e5fefee111e 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -154,7 +154,7 @@ static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream
 	return bytes_to_frames(substream->runtime, iprtd->offset);
 }
 
-static struct snd_pcm_hardware snd_imx_hardware = {
+static const struct snd_pcm_hardware snd_imx_hardware = {
 	.info = SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		SNDRV_PCM_INFO_MMAP |
-- 
cgit v1.2.3


From 636d7e289c0a59dd3febfe6a2e7e5bd56eaa91b8 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 17 Aug 2017 15:46:10 +0530
Subject: ASoC: kirkwood: make snd_pcm_hardware const

Make this const as it is either passed as the 2nd argument
to the function snd_soc_set_runtime_hwparams, which is const or used in
a copy operation.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/kirkwood/kirkwood-dma.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index dafd22e874e9..cf23af159acf 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -27,7 +27,7 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
 	return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai);
 }
 
-static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
+static const struct snd_pcm_hardware kirkwood_dma_snd_hw = {
 	.info = SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_MMAP_VALID |
-- 
cgit v1.2.3


From 5c2e035e69ffff4c8b3f0a2ddfefa102ad9d9d42 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 17 Aug 2017 15:46:11 +0530
Subject: ASoC: sh: make snd_pcm_hardware const

Make these const as they are only passed as the 2nd argument to the
function snd_soc_set_runtime_hwparams, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/dma-sh7760.c | 2 +-
 sound/soc/sh/fsi.c        | 2 +-
 sound/soc/sh/rcar/core.c  | 2 +-
 sound/soc/sh/siu_dai.c    | 2 +-
 4 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 35788a6db963..1e7d417b53ef 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -89,7 +89,7 @@ struct camelot_pcm {
 #define DMABRG_PREALLOC_BUFFER		32 * 1024
 #define DMABRG_PREALLOC_BUFFER_MAX	32 * 1024
 
-static struct snd_pcm_hardware camelot_pcm_hardware = {
+static const struct snd_pcm_hardware camelot_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 5d7d9fe8bc98..39ec772912d5 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1710,7 +1710,7 @@ static const struct snd_soc_dai_ops fsi_dai_ops = {
  *		pcm ops
  */
 
-static struct snd_pcm_hardware fsi_pcm_hardware = {
+static const struct snd_pcm_hardware fsi_pcm_hardware = {
 	.info =		SNDRV_PCM_INFO_INTERLEAVED	|
 			SNDRV_PCM_INFO_MMAP		|
 			SNDRV_PCM_INFO_MMAP_VALID,
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index df39831b8e8f..107133297e8d 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -843,7 +843,7 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
 				ir, &ic);
 }
 
-static struct snd_pcm_hardware rsnd_pcm_hardware = {
+static const struct snd_pcm_hardware rsnd_pcm_hardware = {
 	.info =		SNDRV_PCM_INFO_INTERLEAVED	|
 			SNDRV_PCM_INFO_MMAP		|
 			SNDRV_PCM_INFO_MMAP_VALID,
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index 4a22aadac294..160502947da2 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -333,7 +333,7 @@ static void siu_dai_spbstop(struct siu_port *port_info)
 /*		API functions		*/
 
 /* Playback and capture hardware properties are identical */
-static struct snd_pcm_hardware siu_dai_pcm_hw = {
+static const struct snd_pcm_hardware siu_dai_pcm_hw = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED,
 	.formats		= SNDRV_PCM_FMTBIT_S16,
 	.rates			= SNDRV_PCM_RATE_8000_48000,
-- 
cgit v1.2.3


From 193e25e1fe822f30a02aa9bf99880345eb242d35 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 17 Aug 2017 15:46:12 +0530
Subject: ASoC: qcom: make snd_pcm_hardware const

Make this const as it is either passed as the 2nd argument
to the function snd_soc_set_runtime_hwparams, which is const or used
in a copy operation.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/lpass-platform.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index fb3576af7911..e1945e1772cd 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -32,7 +32,7 @@ struct lpass_pcm_data {
 #define LPASS_PLATFORM_BUFFER_SIZE	(16 * 1024)
 #define LPASS_PLATFORM_PERIODS		2
 
-static struct snd_pcm_hardware lpass_platform_pcm_hardware = {
+static const struct snd_pcm_hardware lpass_platform_pcm_hardware = {
 	.info			=	SNDRV_PCM_INFO_MMAP |
 					SNDRV_PCM_INFO_MMAP_VALID |
 					SNDRV_PCM_INFO_INTERLEAVED |
-- 
cgit v1.2.3


From a7468e478a95db31c0f64e8497fadd9df5c49789 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 17 Aug 2017 15:46:08 +0530
Subject: ASoC: Intel: Atom: make snd_pcm_hardware const

Make this const as it is only used in a copy operation.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index b272df5ce0e8..43e7fdd19f29 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -76,7 +76,7 @@ int sst_unregister_dsp(struct sst_device *dev)
 }
 EXPORT_SYMBOL_GPL(sst_unregister_dsp);
 
-static struct snd_pcm_hardware sst_platform_pcm_hw = {
+static const struct snd_pcm_hardware sst_platform_pcm_hw = {
 	.info =	(SNDRV_PCM_INFO_INTERLEAVED |
 			SNDRV_PCM_INFO_DOUBLE |
 			SNDRV_PCM_INFO_PAUSE |
-- 
cgit v1.2.3


From 8df397ff0e281d5324d4b7a7e9fa56c4188e0a66 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Thu, 17 Aug 2017 15:46:09 +0530
Subject: ASoC: Intel: Skylake: make snd_pcm_hardware const

Make this const as it is only passed as the 2nd argument to the
function snd_soc_set_runtime_hwparams, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index e98d8252a026..f6c9adb6faa2 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -33,7 +33,7 @@
 #define HDA_STEREO 2
 #define HDA_QUAD 4
 
-static struct snd_pcm_hardware azx_pcm_hw = {
+static const struct snd_pcm_hardware azx_pcm_hw = {
 	.info =			(SNDRV_PCM_INFO_MMAP |
 				 SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-- 
cgit v1.2.3


From 23c7159a799f3364720b3d250f57bc8e26b17287 Mon Sep 17 00:00:00 2001
From: huang lin
Date: Thu, 17 Aug 2017 10:24:44 +0800
Subject: ASoC: codec: use enable pin to control dmic start and stop

on some board use enable pin to control dmic start and stop,
so add this feature in dmic driver.

Signed-off-by: Lin Huang <hl@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig |  2 +-
 sound/soc/codecs/dmic.c  | 48 +++++++++++++++++++++++++++++++++++++++++++++++-
 2 files changed, 48 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 6c78b0b49b81..952c44dd5e90 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
 	select SND_SOC_DIO2125
-	select SND_SOC_DMIC
+	select SND_SOC_DMIC if GPIOLIB
 	select SND_SOC_ES8316 if I2C
 	select SND_SOC_ES8328_SPI if SPI_MASTER
 	select SND_SOC_ES8328_I2C if I2C
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index d50f142e2972..b88a1ee66f80 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -19,6 +19,8 @@
  *
  */
 
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/module.h>
@@ -27,6 +29,34 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
+static int dmic_daiops_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_dai *dai)
+{
+	struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai);
+
+	if (!dmic_en)
+		return 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		gpiod_set_value(dmic_en, 1);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		gpiod_set_value(dmic_en, 0);
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops dmic_dai_ops = {
+	.trigger	= dmic_daiops_trigger,
+};
+
 static struct snd_soc_dai_driver dmic_dai = {
 	.name = "dmic-hifi",
 	.capture = {
@@ -38,8 +68,23 @@ static struct snd_soc_dai_driver dmic_dai = {
 			| SNDRV_PCM_FMTBIT_S24_LE
 			| SNDRV_PCM_FMTBIT_S16_LE,
 	},
+	.ops    = &dmic_dai_ops,
 };
 
+static int dmic_codec_probe(struct snd_soc_codec *codec)
+{
+	struct gpio_desc *dmic_en;
+
+	dmic_en = devm_gpiod_get_optional(codec->dev,
+					"dmicen", GPIOD_OUT_LOW);
+	if (IS_ERR(dmic_en))
+		return PTR_ERR(dmic_en);
+
+	snd_soc_codec_set_drvdata(codec, dmic_en);
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0,
 			     SND_SOC_NOPM, 0, 0),
@@ -50,7 +95,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"DMIC AIF", NULL, "DMic"},
 };
 
-static struct snd_soc_codec_driver soc_dmic = {
+static const struct snd_soc_codec_driver soc_dmic = {
+	.probe = dmic_codec_probe,
 	.component_driver = {
 		.dapm_widgets		= dmic_dapm_widgets,
 		.num_dapm_widgets	= ARRAY_SIZE(dmic_dapm_widgets),
-- 
cgit v1.2.3


From d46b1828406f74bc2d9ffe15e9a7b76ddbffbf7f Mon Sep 17 00:00:00 2001
From: Harsha Priya N
Date: Wed, 16 Aug 2017 16:19:55 +0530
Subject: ASoC: Intel: kbl: Enabling ASRC for RT5663 codec on kabylake platform

Patch fixes cracking noise in rt5663 headphones for kbl platform by
calling rt5663_sel_asrc_clk_src() for RT5663_AD_STEREO_FILTER to set
ASRC.

The ASRC function is for asynchronous MCLK and LRCLK. For RT5663 ASRC
should be enabled to support pcm format with 100fs.
ASRC function will track i2s clock and generate corresponding
system clock for codec. Calling this func helps select clock source
for both RT5663_AD_STEREO_FILTER and RT5663_DA_STEREO_FILTER filters
which fixes the crackling sound.

Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Shruthi Sudhakar <shruthi.sudhakar@intel.com>
Signed-off-by: Naveen M <naveen.m@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 8bdd6131c6ad..c1442147d138 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -356,11 +356,13 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	int ret;
 
-	ret = snd_soc_dai_set_sysclk(codec_dai,
-			RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN);
 	/* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */
-	rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1);
+	rt5663_sel_asrc_clk_src(codec_dai->codec,
+			RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER,
+			RT5663_CLK_SEL_I2S1_ASRC);
 
+	ret = snd_soc_dai_set_sysclk(codec_dai,
+			RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret);
 
-- 
cgit v1.2.3


From 052fba77f57b9f243ad3a45a21c809bddcd03627 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:29:24 +0530
Subject: ASoC: aux1x: make snd_soc_ops const

Make this const as it is only stored in the ops field of a
snd_soc_dai_link structure, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/au1x/db1200.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 5c73061d912a..301e1fc9a377 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -99,7 +99,7 @@ static int db1200_i2s_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static struct snd_soc_ops db1200_i2s_wm8731_ops = {
+static const struct snd_soc_ops db1200_i2s_wm8731_ops = {
 	.startup	= db1200_i2s_startup,
 };
 
-- 
cgit v1.2.3


From 7f8159acecb3bb58c0d782813dc605ebef63b317 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:29:25 +0530
Subject: ASoC: cirrus: make snd_soc_ops const

Make these const as they are only stored in the ops field of a
snd_soc_dai_link structure, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/cirrus/edb93xx.c     | 2 +-
 sound/soc/cirrus/snappercl15.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c
index 85962657aabe..c53bd6f2c2d7 100644
--- a/sound/soc/cirrus/edb93xx.c
+++ b/sound/soc/cirrus/edb93xx.c
@@ -56,7 +56,7 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream,
 				      SND_SOC_CLOCK_OUT);
 }
 
-static struct snd_soc_ops edb93xx_ops = {
+static const struct snd_soc_ops edb93xx_ops = {
 	.hw_params	= edb93xx_hw_params,
 };
 
diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c
index 98089df08df6..2334ec19e7eb 100644
--- a/sound/soc/cirrus/snappercl15.c
+++ b/sound/soc/cirrus/snappercl15.c
@@ -45,7 +45,7 @@ static int snappercl15_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops snappercl15_ops = {
+static const struct snd_soc_ops snappercl15_ops = {
 	.hw_params	= snappercl15_hw_params,
 };
 
-- 
cgit v1.2.3


From e23d834876390d6b8d4d35057e9ecd899a44193a Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:29:26 +0530
Subject: ASoC: generic: make snd_soc_ops const

Make these const as they are only stored in the ops field of a
snd_soc_dai_link structure, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card.c     | 2 +-
 sound/soc/generic/audio-graph-scu-card.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 105ec3a6e30d..d3497a3e14ca 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -95,7 +95,7 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
 	asoc_simple_card_clk_disable(&dai_props->codec_dai);
 }
 
-static struct snd_soc_ops asoc_graph_card_ops = {
+static const struct snd_soc_ops asoc_graph_card_ops = {
 	.startup = asoc_graph_card_startup,
 	.shutdown = asoc_graph_card_shutdown,
 };
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index dcd2df37bc3b..3da9acd6844e 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -56,7 +56,7 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
 	asoc_simple_card_clk_disable(dai_props);
 }
 
-static struct snd_soc_ops asoc_graph_card_ops = {
+static const struct snd_soc_ops asoc_graph_card_ops = {
 	.startup = asoc_graph_card_startup,
 	.shutdown = asoc_graph_card_shutdown,
 };
-- 
cgit v1.2.3


From c12a4a40b4fdaefa6c2578e86d277410e1a9cea1 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:29:28 +0530
Subject: ASoC: rockchip: make snd_soc_ops const

Make this const as it is only stored in the ops field of a
snd_soc_dai_link structure, which is const.
Done using Coccinelle

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c
index dbc53e48c52c..42227b0f66ac 100644
--- a/sound/soc/rockchip/rk3288_hdmi_analog.c
+++ b/sound/soc/rockchip/rk3288_hdmi_analog.c
@@ -147,7 +147,7 @@ static int rk_init(struct snd_soc_pcm_runtime *runtime)
 	return 0;
 }
 
-static struct snd_soc_ops rk_ops = {
+static const struct snd_soc_ops rk_ops = {
 	.hw_params = rk_hw_params,
 };
 
-- 
cgit v1.2.3


From 2af236391f5a167630ebca0eced832afcc4634e0 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:29:29 +0530
Subject: ASoC: samsung: make snd_soc_ops const

Make these const as they are only stored in the ops field of a
snd_soc_dai_link structure, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/jive_wm8750.c     | 2 +-
 sound/soc/samsung/s3c24xx_simtec.c  | 2 +-
 sound/soc/samsung/s3c24xx_uda134x.c | 2 +-
 sound/soc/samsung/smdk_spdif.c      | 2 +-
 4 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 7fcb51faa2a0..529b10dc532b 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -79,7 +79,7 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops jive_ops = {
+static const struct snd_soc_ops jive_ops = {
 	.hw_params	= jive_hw_params,
 };
 
diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c
index dcc008d1e1ab..6de63f3e37d5 100644
--- a/sound/soc/samsung/s3c24xx_simtec.c
+++ b/sound/soc/samsung/s3c24xx_simtec.c
@@ -211,7 +211,7 @@ static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd)
 	return 0;
 }
 
-static struct snd_soc_ops simtec_snd_ops = {
+static const struct snd_soc_ops simtec_snd_ops = {
 	.hw_params	= simtec_hw_params,
 };
 
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index 55538e333cc8..c65c89d4aaa6 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -199,7 +199,7 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops s3c24xx_uda134x_ops = {
+static const struct snd_soc_ops s3c24xx_uda134x_ops = {
 	.startup = s3c24xx_uda134x_startup,
 	.shutdown = s3c24xx_uda134x_shutdown,
 	.hw_params = s3c24xx_uda134x_hw_params,
diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c
index a2f2363fe1c2..7fc7cc6d1530 100644
--- a/sound/soc/samsung/smdk_spdif.c
+++ b/sound/soc/samsung/smdk_spdif.c
@@ -144,7 +144,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static struct snd_soc_ops smdk_spdif_ops = {
+static const struct snd_soc_ops smdk_spdif_ops = {
 	.hw_params = smdk_hw_params,
 };
 
-- 
cgit v1.2.3


From a46d0592d164c017342d2d86e5d123735ec046ae Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:29:30 +0530
Subject: ASoC: sh: make snd_soc_ops const

Make this const as it is only stored in the ops field of a
snd_soc_dai_link structure, which is const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/migor.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index 672bcd4c252b..ecb057ff9fbb 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -98,7 +98,7 @@ static int migor_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static struct snd_soc_ops migor_dai_ops = {
+static const struct snd_soc_ops migor_dai_ops = {
 	.hw_params = migor_hw_params,
 	.hw_free = migor_hw_free,
 };
-- 
cgit v1.2.3


From cc6e25631ddd4f3295fc79568f2145c437849e63 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:45:08 +0530
Subject: ASoC: bcm: make snd_soc_dai_driver const

Make this const as it is only used during a copy operation.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/bcm/cygnus-ssp.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index e710bb0c5637..cdd5e82439dd 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1288,7 +1288,7 @@ static const struct snd_soc_dai_driver cygnus_ssp_dai_info[] = {
 	INIT_CPU_DAI(2),
 };
 
-static struct snd_soc_dai_driver cygnus_spdif_dai_info = {
+static const struct snd_soc_dai_driver cygnus_spdif_dai_info = {
 	.name = "cygnus-spdif",
 	.playback = {
 		.channels_min = 2,
-- 
cgit v1.2.3


From 6e37f933ed1ade3201dd609d103c0ef79c0ef6b0 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 16 Aug 2017 22:45:09 +0530
Subject: ASoC: codecs: make snd_soc_dai_driver and snd_soc_component_driver
 const

Make these two structure variables const as they are either used in a
copy operation or passed to devm_snd_soc_register_component having the
corresponding argument as const.
Done using Coccinelle.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdmi-codec.c | 2 +-
 sound/soc/codecs/lm4857.c     | 2 +-
 sound/soc/codecs/max9768.c    | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index f288404f0a61..d73d2c1ed823 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -692,7 +692,7 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 	return snd_ctl_add(rtd->card->snd_card, kctl);
 }
 
-static struct snd_soc_dai_driver hdmi_i2s_dai = {
+static const struct snd_soc_dai_driver hdmi_i2s_dai = {
 	.name = "i2s-hifi",
 	.id = DAI_ID_I2S,
 	.playback = {
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 558de1053f73..1e964079642a 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -100,7 +100,7 @@ static const struct snd_soc_dapm_route lm4857_routes[] = {
 	{ "EP", "Earpiece", "Mode" },
 };
 
-static struct snd_soc_component_driver lm4857_component_driver = {
+static const struct snd_soc_component_driver lm4857_component_driver = {
 	.controls = lm4857_controls,
 	.num_controls = ARRAY_SIZE(lm4857_controls),
 	.dapm_widgets = lm4857_dapm_widgets,
diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c
index 5b82e26cd5d1..7017c0389e73 100644
--- a/sound/soc/codecs/max9768.c
+++ b/sound/soc/codecs/max9768.c
@@ -151,7 +151,7 @@ static int max9768_probe(struct snd_soc_component *component)
 	return 0;
 }
 
-static struct snd_soc_component_driver max9768_component_driver = {
+static const struct snd_soc_component_driver max9768_component_driver = {
 	.probe = max9768_probe,
 	.controls = max9768_volume,
 	.num_controls = ARRAY_SIZE(max9768_volume),
-- 
cgit v1.2.3


From 3bcfe80f1c471b6bbade9f1a725778ea0471bb49 Mon Sep 17 00:00:00 2001
From: Paul Kocialkowski
Date: Thu, 17 Aug 2017 21:05:52 +0200
Subject: ASoC: tegra: Remove SoC-specific Kconfig depends and selects

This removes the SoC-specific dependencies on the platform drivers,
as well as SoC-specific selections of platform drivers for the
machine drivers. The rationale behind this change is that the
dependencies are not actual build dependencies but run-time ones.

The previously listed SoCs were also incomplete: for instance, tegra124
uses the tegra30 platform drivers, which could not be built without
ARCH_TEGRA_3x_SOC set.

Descriptions of the tristates are also added to allow these options
to be set via defconfig/menuconfig.

Signed-off-by: Paul Kocialkowski <contact@paulk.fr>
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/Kconfig | 42 ++++++++++++++----------------------------
 1 file changed, 14 insertions(+), 28 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index efbe8d4c019e..6875fc39a575 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -9,8 +9,8 @@ config SND_SOC_TEGRA
 	  Say Y or M here if you want support for SoC audio on Tegra.
 
 config SND_SOC_TEGRA20_AC97
-	tristate
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+	tristate "Tegra20 AC97 interface"
+	depends on SND_SOC_TEGRA
 	select SND_SOC_AC97_BUS
 	select SND_SOC_TEGRA20_DAS
 	help
@@ -19,16 +19,16 @@ config SND_SOC_TEGRA20_AC97
 	  machine drivers to support below.
 
 config SND_SOC_TEGRA20_DAS
-	tristate
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+	tristate "Tegra20 DAS module"
+	depends on SND_SOC_TEGRA
 	help
 	  Say Y or M if you want to add support for the Tegra20 DAS module.
 	  You will also need to select the individual machine drivers to
 	  support below.
 
 config SND_SOC_TEGRA20_I2S
-	tristate
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+	tristate "Tegra20 I2S interface"
+	depends on SND_SOC_TEGRA
 	select SND_SOC_TEGRA20_DAS
 	help
 	  Say Y or M if you want to add support for codecs attached to the
@@ -36,8 +36,8 @@ config SND_SOC_TEGRA20_I2S
 	  machine drivers to support below.
 
 config SND_SOC_TEGRA20_SPDIF
-	tristate
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+	tristate "Tegra20 SPDIF interface"
+	depends on SND_SOC_TEGRA
 	default m
 	help
 	  Say Y or M if you want to add support for the Tegra20 SPDIF interface.
@@ -45,16 +45,16 @@ config SND_SOC_TEGRA20_SPDIF
 	  below.
 
 config SND_SOC_TEGRA30_AHUB
-	tristate
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC
+	tristate "Tegra30 AHUB module"
+	depends on SND_SOC_TEGRA
 	help
-	  Say Y or M if you want to add support for the Tegra20 AHUB module.
+	  Say Y or M if you want to add support for the Tegra30 AHUB module.
 	  You will also need to select the individual machine drivers to
 	  support below.
 
 config SND_SOC_TEGRA30_I2S
-	tristate
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC
+	tristate "Tegra30 I2S interface"
+	depends on SND_SOC_TEGRA
 	select SND_SOC_TEGRA30_AHUB
 	help
 	  Say Y or M if you want to add support for codecs attached to the
@@ -64,8 +64,6 @@ config SND_SOC_TEGRA30_I2S
 config SND_SOC_TEGRA_RT5640
 	tristate "SoC Audio support for Tegra boards using an RT5640 codec"
 	depends on SND_SOC_TEGRA && I2C && GPIOLIB
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
-	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_RT5640
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
@@ -74,8 +72,6 @@ config SND_SOC_TEGRA_RT5640
 config SND_SOC_TEGRA_WM8753
 	tristate "SoC Audio support for Tegra boards using a WM8753 codec"
 	depends on SND_SOC_TEGRA && I2C && GPIOLIB
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
-	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_WM8753
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
@@ -84,8 +80,6 @@ config SND_SOC_TEGRA_WM8753
 config SND_SOC_TEGRA_WM8903
 	tristate "SoC Audio support for Tegra boards using a WM8903 codec"
 	depends on SND_SOC_TEGRA && I2C && GPIOLIB
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
-	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_WM8903
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
@@ -94,7 +88,7 @@ config SND_SOC_TEGRA_WM8903
 
 config SND_SOC_TEGRA_WM9712
 	tristate "SoC Audio support for Tegra boards using a WM9712 codec"
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB
+	depends on SND_SOC_TEGRA && GPIOLIB
 	select SND_SOC_TEGRA20_AC97
 	select SND_SOC_WM9712
 	help
@@ -104,7 +98,6 @@ config SND_SOC_TEGRA_WM9712
 config SND_SOC_TEGRA_TRIMSLICE
 	tristate "SoC Audio support for TrimSlice board"
 	depends on SND_SOC_TEGRA && I2C
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
 	select SND_SOC_TLV320AIC23_I2C
 	help
 	  Say Y or M here if you want to add support for SoC audio on the
@@ -113,7 +106,6 @@ config SND_SOC_TEGRA_TRIMSLICE
 config SND_SOC_TEGRA_ALC5632
 	tristate "SoC Audio support for Tegra boards using an ALC5632 codec"
 	depends on SND_SOC_TEGRA && I2C && GPIOLIB
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
 	select SND_SOC_ALC5632
 	help
 	  Say Y or M here if you want to add support for SoC audio on the
@@ -122,8 +114,6 @@ config SND_SOC_TEGRA_ALC5632
 config SND_SOC_TEGRA_MAX98090
 	tristate "SoC Audio support for Tegra boards using a MAX98090 codec"
 	depends on SND_SOC_TEGRA && I2C && GPIOLIB
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
-	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_MAX98090
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
@@ -132,8 +122,6 @@ config SND_SOC_TEGRA_MAX98090
 config SND_SOC_TEGRA_RT5677
 	tristate "SoC Audio support for Tegra boards using a RT5677 codec"
 	depends on SND_SOC_TEGRA && I2C && GPIOLIB
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
-	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_RT5677
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
@@ -142,8 +130,6 @@ config SND_SOC_TEGRA_RT5677
 config SND_SOC_TEGRA_SGTL5000
 	tristate "SoC Audio support for Tegra boards using a SGTL5000 codec"
 	depends on SND_SOC_TEGRA && I2C && GPIOLIB
-	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
-	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_SGTL5000
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
-- 
cgit v1.2.3


From f6c1626e28f22115ded520300b56fa049c068f4f Mon Sep 17 00:00:00 2001
From: Ryder Lee
Date: Fri, 18 Aug 2017 09:28:50 +0800
Subject: ASoC: mediatek: switch to use platform_get_irq_byname()

Switch to use platform_get_irq_byname() so that binding can be agnostic
of the IRQ order.

Signed-off-by: Ryder Lee <ryder.lee@mediatek.com>
CC: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 13 ++++++-------
 1 file changed, 6 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
index 10dcd69c5ffa..8fda182f849b 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
@@ -1496,14 +1496,12 @@ static int mt2701_afe_runtime_resume(struct device *dev)
 
 static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev)
 {
-	int ret, i;
-	unsigned int irq_id;
 	struct mtk_base_afe *afe;
 	struct mt2701_afe_private *afe_priv;
 	struct resource *res;
 	struct device *dev;
+	int i, irq_id, ret;
 
-	ret = 0;
 	afe = devm_kzalloc(&pdev->dev, sizeof(*afe), GFP_KERNEL);
 	if (!afe)
 		return -ENOMEM;
@@ -1516,11 +1514,12 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev)
 	afe->dev = &pdev->dev;
 	dev = afe->dev;
 
-	irq_id = platform_get_irq(pdev, 0);
-	if (!irq_id) {
-		dev_err(dev, "%s no irq found\n", dev->of_node->name);
-		return -ENXIO;
+	irq_id = platform_get_irq_byname(pdev, "asys");
+	if (irq_id < 0) {
+		dev_err(dev, "unable to get ASYS IRQ\n");
+		return irq_id;
 	}
+
 	ret = devm_request_irq(dev, irq_id, mt2701_asys_isr,
 			       IRQF_TRIGGER_NONE, "asys-isr", (void *)afe);
 	if (ret) {
-- 
cgit v1.2.3


From 728c4a87f31d30669a103dac1851a15f4e2af03c Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Fri, 18 Aug 2017 17:35:59 +0530
Subject: ASoC: codecs: constify snd_soc_dai_ops structures

snd_soc_dai_ops are not supposed to change at runtime. All functions
working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with
const snd_soc_dai_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c             | 2 +-
 sound/soc/codecs/es8316.c              | 2 +-
 sound/soc/codecs/hdac_hdmi.c           | 2 +-
 sound/soc/codecs/inno_rk3036.c         | 2 +-
 sound/soc/codecs/max9867.c             | 2 +-
 sound/soc/codecs/max98926.c            | 2 +-
 sound/soc/codecs/msm8916-wcd-analog.c  | 2 +-
 sound/soc/codecs/msm8916-wcd-digital.c | 2 +-
 sound/soc/codecs/rt5616.c              | 2 +-
 sound/soc/codecs/rt5663.c              | 2 +-
 sound/soc/codecs/tas5720.c             | 2 +-
 sound/soc/codecs/zx_aud96p22.c         | 2 +-
 12 files changed, 12 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 55e4520cdcaf..9e860dfa0163 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -911,7 +911,7 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute)
 			SNDRV_PCM_FMTBIT_S32_LE)
 
 
-static struct snd_soc_dai_ops cs42l42_ops = {
+static const struct snd_soc_dai_ops cs42l42_ops = {
 	.hw_params	= cs42l42_pcm_hw_params,
 	.set_fmt	= cs42l42_set_dai_fmt,
 	.set_sysclk	= cs42l42_set_sysclk,
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index ecc02449c569..f5cf48c9c3b0 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -502,7 +502,7 @@ static int es8316_mute(struct snd_soc_dai *dai, int mute)
 #define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops es8316_ops = {
+static const struct snd_soc_dai_ops es8316_ops = {
 	.startup = es8316_pcm_startup,
 	.hw_params = es8316_pcm_hw_params,
 	.set_fmt = es8316_set_dai_fmt,
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index bc2e74ff3b2d..e6de50acefd4 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1360,7 +1360,7 @@ static void hdac_hdmi_skl_enable_dp12(struct hdac_device *hdac)
 
 }
 
-static struct snd_soc_dai_ops hdmi_dai_ops = {
+static const struct snd_soc_dai_ops hdmi_dai_ops = {
 	.startup = hdac_hdmi_pcm_open,
 	.shutdown = hdac_hdmi_pcm_close,
 	.hw_params = hdac_hdmi_set_hw_params,
diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c
index b918ba5c8ce5..edf42d252fa1 100644
--- a/sound/soc/codecs/inno_rk3036.c
+++ b/sound/soc/codecs/inno_rk3036.c
@@ -310,7 +310,7 @@ static int rk3036_codec_dai_hw_params(struct snd_pcm_substream *substream,
 			   SNDRV_PCM_FMTBIT_S24_LE  | \
 			   SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops rk3036_codec_dai_ops = {
+static const struct snd_soc_dai_ops rk3036_codec_dai_ops = {
 	.set_fmt	= rk3036_codec_dai_set_fmt,
 	.hw_params	= rk3036_codec_dai_hw_params,
 };
diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index 2a40a69a7513..3e421e9d14a5 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -350,7 +350,7 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static struct snd_soc_dai_ops max9867_dai_ops = {
+static const struct snd_soc_dai_ops max9867_dai_ops = {
 	.set_fmt = max9867_dai_set_fmt,
 	.set_sysclk	= max9867_set_dai_sysclk,
 	.prepare	= max9867_prepare,
diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
index 1eff7e0b092e..f16145d8ee6b 100644
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -459,7 +459,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream,
 #define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops max98926_dai_ops = {
+static const struct snd_soc_dai_ops max98926_dai_ops = {
 	.set_fmt = max98926_dai_set_fmt,
 	.hw_params = max98926_dai_hw_params,
 };
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index a78802920c3c..aec1e1626993 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -753,7 +753,7 @@ static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream,
 			    RST_CTL_DIG_SW_RST_N_MASK, 0);
 }
 
-static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = {
+static const struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = {
 	.startup = pm8916_wcd_analog_startup,
 	.shutdown = pm8916_wcd_analog_shutdown,
 };
diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index f690442af8c9..7e3794fb8c2c 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -788,7 +788,7 @@ static void msm8916_wcd_digital_shutdown(struct snd_pcm_substream *substream,
 			    LPASS_CDC_CLK_PDM_CTL_PDM_CLK_SEL_MASK, 0);
 }
 
-static struct snd_soc_dai_ops msm8916_wcd_digital_dai_ops = {
+static const struct snd_soc_dai_ops msm8916_wcd_digital_dai_ops = {
 	.startup = msm8916_wcd_digital_startup,
 	.shutdown = msm8916_wcd_digital_shutdown,
 	.hw_params = msm8916_wcd_digital_hw_params,
diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c
index 7d6e0823f98f..33bdfe7c394a 100644
--- a/sound/soc/codecs/rt5616.c
+++ b/sound/soc/codecs/rt5616.c
@@ -1265,7 +1265,7 @@ static int rt5616_resume(struct snd_soc_codec *codec)
 #define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-static struct snd_soc_dai_ops rt5616_aif_dai_ops = {
+static const struct snd_soc_dai_ops rt5616_aif_dai_ops = {
 	.hw_params = rt5616_hw_params,
 	.set_fmt = rt5616_set_dai_fmt,
 	.set_sysclk = rt5616_set_dai_sysclk,
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index a33202affeb1..af642c0fa280 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -2860,7 +2860,7 @@ static int rt5663_resume(struct snd_soc_codec *codec)
 #define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-static struct snd_soc_dai_ops rt5663_aif_dai_ops = {
+static const struct snd_soc_dai_ops rt5663_aif_dai_ops = {
 	.hw_params = rt5663_hw_params,
 	.set_fmt = rt5663_set_dai_fmt,
 	.set_sysclk = rt5663_set_dai_sysclk,
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index c65b917598d2..01e9ed242378 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -507,7 +507,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
 #define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\
 			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
+static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
 	.hw_params	= tas5720_hw_params,
 	.set_fmt	= tas5720_set_dai_fmt,
 	.set_tdm_slot	= tas5720_set_dai_tdm_slot,
diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c
index 032fb7cf6cbd..855011e9e4d1 100644
--- a/sound/soc/codecs/zx_aud96p22.c
+++ b/sound/soc/codecs/zx_aud96p22.c
@@ -312,7 +312,7 @@ static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static struct snd_soc_dai_ops aud96p22_dai_ops = {
+static const struct snd_soc_dai_ops aud96p22_dai_ops = {
 	.set_fmt = aud96p22_set_fmt,
 };
 
-- 
cgit v1.2.3


From eb59d73cb535ba32df928f210fb9a8529bf465c0 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Fri, 18 Aug 2017 17:35:59 +0530
Subject: ASoC: codecs: constify snd_soc_dai_ops structures

snd_soc_dai_ops are not supposed to change at runtime. All functions
working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with
const snd_soc_dai_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c            | 2 +-
 sound/soc/codecs/es8316.c             | 2 +-
 sound/soc/codecs/inno_rk3036.c        | 2 +-
 sound/soc/codecs/max9867.c            | 2 +-
 sound/soc/codecs/max98926.c           | 2 +-
 sound/soc/codecs/msm8916-wcd-analog.c | 2 +-
 sound/soc/codecs/rt5616.c             | 2 +-
 sound/soc/codecs/rt5663.c             | 2 +-
 sound/soc/codecs/tas5720.c            | 2 +-
 sound/soc/codecs/zx_aud96p22.c        | 2 +-
 10 files changed, 10 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 55e4520cdcaf..9e860dfa0163 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -911,7 +911,7 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute)
 			SNDRV_PCM_FMTBIT_S32_LE)
 
 
-static struct snd_soc_dai_ops cs42l42_ops = {
+static const struct snd_soc_dai_ops cs42l42_ops = {
 	.hw_params	= cs42l42_pcm_hw_params,
 	.set_fmt	= cs42l42_set_dai_fmt,
 	.set_sysclk	= cs42l42_set_sysclk,
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 4f35af6a5d3a..da2d353af5ba 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -502,7 +502,7 @@ static int es8316_mute(struct snd_soc_dai *dai, int mute)
 #define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops es8316_ops = {
+static const struct snd_soc_dai_ops es8316_ops = {
 	.startup = es8316_pcm_startup,
 	.hw_params = es8316_pcm_hw_params,
 	.set_fmt = es8316_set_dai_fmt,
diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c
index 64b0be94bea3..6b59b6f08298 100644
--- a/sound/soc/codecs/inno_rk3036.c
+++ b/sound/soc/codecs/inno_rk3036.c
@@ -310,7 +310,7 @@ static int rk3036_codec_dai_hw_params(struct snd_pcm_substream *substream,
 			   SNDRV_PCM_FMTBIT_S24_LE  | \
 			   SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops rk3036_codec_dai_ops = {
+static const struct snd_soc_dai_ops rk3036_codec_dai_ops = {
 	.set_fmt	= rk3036_codec_dai_set_fmt,
 	.hw_params	= rk3036_codec_dai_hw_params,
 };
diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index 6c0c0d6e8f3c..2f60924fe919 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -350,7 +350,7 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static struct snd_soc_dai_ops max9867_dai_ops = {
+static const struct snd_soc_dai_ops max9867_dai_ops = {
 	.set_fmt = max9867_dai_set_fmt,
 	.set_sysclk	= max9867_set_dai_sysclk,
 	.prepare	= max9867_prepare,
diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
index 7a39bfb9e0f9..59d03d036945 100644
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -459,7 +459,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream,
 #define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-static struct snd_soc_dai_ops max98926_dai_ops = {
+static const struct snd_soc_dai_ops max98926_dai_ops = {
 	.set_fmt = max98926_dai_set_fmt,
 	.hw_params = max98926_dai_hw_params,
 };
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index 4cc52cecf3b7..f07674eddc52 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -753,7 +753,7 @@ static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream,
 			    RST_CTL_DIG_SW_RST_N_MASK, 0);
 }
 
-static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = {
+static const struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = {
 	.startup = pm8916_wcd_analog_startup,
 	.shutdown = pm8916_wcd_analog_shutdown,
 };
diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c
index 3c5f555ae6d0..c94e94fe8297 100644
--- a/sound/soc/codecs/rt5616.c
+++ b/sound/soc/codecs/rt5616.c
@@ -1265,7 +1265,7 @@ static int rt5616_resume(struct snd_soc_codec *codec)
 #define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-static struct snd_soc_dai_ops rt5616_aif_dai_ops = {
+static const struct snd_soc_dai_ops rt5616_aif_dai_ops = {
 	.hw_params = rt5616_hw_params,
 	.set_fmt = rt5616_set_dai_fmt,
 	.set_sysclk = rt5616_set_dai_sysclk,
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index ebc5e3b2b62b..1bbe36338cd3 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -2860,7 +2860,7 @@ static int rt5663_resume(struct snd_soc_codec *codec)
 #define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-static struct snd_soc_dai_ops rt5663_aif_dai_ops = {
+static const struct snd_soc_dai_ops rt5663_aif_dai_ops = {
 	.hw_params = rt5663_hw_params,
 	.set_fmt = rt5663_set_dai_fmt,
 	.set_sysclk = rt5663_set_dai_sysclk,
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index d14b78b1917b..a736a2a6976c 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -507,7 +507,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
 #define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\
 			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
 
-static struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
+static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
 	.hw_params	= tas5720_hw_params,
 	.set_fmt	= tas5720_set_dai_fmt,
 	.set_tdm_slot	= tas5720_set_dai_tdm_slot,
diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c
index 36d0a7087a54..51fad48892d3 100644
--- a/sound/soc/codecs/zx_aud96p22.c
+++ b/sound/soc/codecs/zx_aud96p22.c
@@ -312,7 +312,7 @@ static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static struct snd_soc_dai_ops aud96p22_dai_ops = {
+static const struct snd_soc_dai_ops aud96p22_dai_ops = {
 	.set_fmt = aud96p22_set_fmt,
 };
 
-- 
cgit v1.2.3


From ab566fd52309971da595af057fd480aed82ca0e2 Mon Sep 17 00:00:00 2001
From: Arvind Yadav
Date: Fri, 18 Aug 2017 17:36:10 +0530
Subject: ASoC: ux500: constify snd_soc_dai_ops structures

snd_soc_dai_ops are not supposed to change at runtime. All functions
working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with
const snd_soc_dai_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ux500/ux500_msp_dai.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index ec5152aa3f6e..625b72a5facd 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -707,7 +707,7 @@ static int ux500_msp_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops ux500_msp_dai_ops[] = {
+static const struct snd_soc_dai_ops ux500_msp_dai_ops[] = {
 	{
 		.set_sysclk = ux500_msp_dai_set_dai_sysclk,
 		.set_fmt = ux500_msp_dai_set_dai_fmt,
-- 
cgit v1.2.3


From b65cb7a590274bf5dbf5bbccbbd0c495c27ab14d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Mon, 21 Aug 2017 07:03:01 +0000
Subject: ASoC: rsnd: tidyup comments position/space/tab

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/dma.c | 2 +-
 sound/soc/sh/rcar/src.c | 7 +++----
 2 files changed, 4 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index 60aa5e96a49f..041ec1080d52 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -604,8 +604,8 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
 		dma_addr_t in_addr;
 	} dma_addrs[3][2][3] = {
 		/* SRC */
+		/* Capture */
 		{{{ 0,				0 },
-		  /* Capture */
 		  { RDMA_SRC_O_N(src, id),	RDMA_SRC_I_P(src, id) },
 		  { RDMA_CMD_O_N(src, id),	RDMA_SRC_I_P(src, id) } },
 		 /* Playback */
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 8e9a1de0ec25..46400397dbbd 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -108,7 +108,6 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv,
 	int is_play = rsnd_io_is_play(io);
 
 	/*
-	 *
 	 * Playback
 	 * runtime_rate -> [SRC] -> convert_rate
 	 *
@@ -203,13 +202,13 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
 	use_src = (fin != fout) | rsnd_src_sync_is_enabled(mod);
 
 	/*
-	 *	SRC_ADINR
+	 * SRC_ADINR
 	 */
 	adinr = rsnd_get_adinr_bit(mod, io) |
 		rsnd_runtime_channel_original(io);
 
 	/*
-	 *	SRC_IFSCR / SRC_IFSVR
+	 * SRC_IFSCR / SRC_IFSVR
 	 */
 	ifscr = 0;
 	fsrate = 0;
@@ -223,7 +222,7 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io,
 	}
 
 	/*
-	 *	SRC_SRCCR / SRC_ROUTE_MODE0
+	 * SRC_SRCCR / SRC_ROUTE_MODE0
 	 */
 	cr	= 0x00011110;
 	route	= 0x0;
-- 
cgit v1.2.3


From 1107a293b7c33723ace2a8d18f5c6febcd576d7a Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:21:35 +0800
Subject: ASoC: s3c24xx_uda134x: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to card->dev,then
the function platform_set_drvdata copies the value the variable card
to pdev->dev.driver_data, but when calling snd_soc_register_card,the
function dev_set_drvdata(card->dev, card) will also do the same copy
operation,so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/s3c24xx_uda134x.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index c65c89d4aaa6..5fb3bab6bbfe 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -237,7 +237,6 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev)
 	mutex_init(&priv->clk_lock);
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, priv);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-- 
cgit v1.2.3


From 5c23bc7cb6b6b40ddf7864e5dead647625e2f386 Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:23:19 +0800
Subject: ASoC: atmel: Remove unnecessary function call

The function platform_set_drvdata copies the value the variable
card to card->dev->driver_data, then the address of &pdev->dev is
assigned to card->dev in atmel_classd_asoc_card_init, but when
calling snd_soc_register_card,the function
dev_set_drvdata(card->dev, card) will do the same copy operation,
so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/atmel-classd.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index b7ef8c59b49a..0cd7caaed9c4 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -652,7 +652,6 @@ static int atmel_classd_probe(struct platform_device *pdev)
 	}
 
 	snd_soc_card_set_drvdata(card, dd);
-	platform_set_drvdata(pdev, card);
 
 	ret = atmel_classd_asoc_card_init(dev, card);
 	if (ret) {
-- 
cgit v1.2.3


From acd617a461ed811894f4318c9b952fa002f3be21 Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:25:25 +0800
Subject: ASoC: atmel: Remove unnecessary function call

The function platform_set_drvdata copies the value the variable
card to card->dev->driver_data, then the address of &pdev->dev is
assigned to card->dev in atmel_pdmic_asoc_card_init, but when
calling snd_soc_register_card,the function
dev_set_drvdata(card->dev, card) will do the same copy operation,
so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/atmel-pdmic.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c
index c917df7715d1..8e3d34be9e69 100644
--- a/sound/soc/atmel/atmel-pdmic.c
+++ b/sound/soc/atmel/atmel-pdmic.c
@@ -694,7 +694,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev)
 	}
 
 	snd_soc_card_set_drvdata(card, dd);
-	platform_set_drvdata(pdev, card);
 
 	ret = atmel_pdmic_asoc_card_init(dev, card);
 	if (ret) {
-- 
cgit v1.2.3


From dc94d245a57b76f38c7a936cf900cc38237638de Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:25:46 +0800
Subject: ASoC: rockchip: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to card->dev,then
the function platform_set_drvdata copies the value the variable card
to pdev->dev.driver_data, but when calling snd_soc_register_card,the
function dev_set_drvdata(card->dev, card) will also do the same copy
operation,so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rk3288_hdmi_analog.c | 2 --
 sound/soc/rockchip/rk3399_gru_sound.c   | 1 -
 2 files changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c
index 42227b0f66ac..fa44e3901336 100644
--- a/sound/soc/rockchip/rk3288_hdmi_analog.c
+++ b/sound/soc/rockchip/rk3288_hdmi_analog.c
@@ -272,8 +272,6 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	platform_set_drvdata(pdev, card);
-
 	return ret;
 }
 
diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index 3475c61a5fa0..70c0908fb9b4 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -362,7 +362,6 @@ static int rockchip_sound_probe(struct platform_device *pdev)
 	rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL);
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
-- 
cgit v1.2.3


From 4c316783ee88eee4f045fbb31344471c6607b6ed Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:26:25 +0800
Subject: ASoC: mxs-sgtl5000: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to card->dev,then
the function platform_set_drvdata copies the value the variable card
to pdev->dev.driver_data, but when calling snd_soc_register_card,the
function dev_set_drvdata(card->dev, card) will also do the same copy
operation,so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mxs/mxs-sgtl5000.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index a96276e77332..2ed3240cc682 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -140,7 +140,6 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
 	}
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
-- 
cgit v1.2.3


From 70024ebc51cc4a10a2945188e1129ae594830ca2 Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:43:57 +0800
Subject: ASoC: qcom: Remove useless function call

The function platform_set_drvdata(pdev, data) copies the value of
the variable data to pdev->dev.driver_data,but when calling
snd_soc_register_card,the function dev_set_drvdata(card->dev, card)
will override it, so i think that the former copy operation is
useless and can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Acked-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/apq8016_sbc.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index f07aa1e1cdfe..96a079d9f697 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -206,7 +206,6 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev)
 	if (IS_ERR(data->spkr_iomux))
 		return PTR_ERR(data->spkr_iomux);
 
-	platform_set_drvdata(pdev, data);
 	snd_soc_card_set_drvdata(card, data);
 
 	return devm_snd_soc_register_card(&pdev->dev, card);
-- 
cgit v1.2.3


From 3bb2991c1d4d487323fde7b5b63985798c91a03f Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:44:58 +0800
Subject: ASoC: qcom: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to card->dev,then
the function platform_set_drvdata copies the value the variable card
to pdev->dev.driver_data, but when calling snd_soc_register_card,the
function dev_set_drvdata(card->dev, card) will also do the same copy
operation,so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Acked-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/storm.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c
index c5207af14104..a9fa972466ad 100644
--- a/sound/soc/qcom/storm.c
+++ b/sound/soc/qcom/storm.c
@@ -99,7 +99,6 @@ static int storm_platform_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 
 	ret = snd_soc_of_parse_card_name(card, "qcom,model");
 	if (ret) {
-- 
cgit v1.2.3


From 827807549e690c84bba5aac99b58dfbd52fcef8c Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:55:13 +0800
Subject: ASoC: sun4i-codec: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to card->dev in
create_card,then the function platform_set_drvdata copies the value
the variable card to pdev->dev.driver_data, but when calling
snd_soc_register_card,the function dev_set_drvdata(card->dev, card)
will also do the same copy operation,so i think that the former copy
operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-codec.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index c668b8b08b38..baa9007464ed 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -1655,7 +1655,6 @@ static int sun4i_codec_probe(struct platform_device *pdev)
 		goto err_unregister_codec;
 	}
 
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, scodec);
 
 	ret = snd_soc_register_card(card);
-- 
cgit v1.2.3


From 6eb4f27419d9250ac632df15d9dcf916d84b9944 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:32 +0200
Subject: ASoC: sun4i-i2s: Add regmap fields for channels

On the original i2s block the channel mapping and selection were
configured for stereo audio by default: This is not the case with
the newer SoCs and they are also located at different offsets.

To support the newer SoC then regmap fields have been added to the
quirks and these are initialised to their correct settings during
probing.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 77 ++++++++++++++++++++++++++++++++++++++++-----
 1 file changed, 69 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index dfb794ffff92..87feb5a14cff 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -82,7 +82,7 @@
 #define SUN4I_I2S_TX_CNT_REG		0x2c
 
 #define SUN4I_I2S_TX_CHAN_SEL_REG	0x30
-#define SUN4I_I2S_TX_CHAN_SEL(num_chan)		(((num_chan) - 1) << 0)
+#define SUN4I_I2S_CHAN_SEL(num_chan)		(((num_chan) - 1) << 0)
 
 #define SUN4I_I2S_TX_CHAN_MAP_REG	0x34
 #define SUN4I_I2S_TX_CHAN_MAP(chan, sample)	((sample) << (chan << 2))
@@ -98,6 +98,10 @@
  * @sun4i_i2s_regmap: regmap config to use.
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
  * @bclk_offset: Value by which bclkdiv needs to be adjusted.
+ * @field_txchanmap: location of the tx channel mapping register.
+ * @field_rxchanmap: location of the rx channel mapping register.
+ * @field_txchansel: location of the tx channel select bit fields.
+ * @field_rxchansel: location of the rx channel select bit fields.
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
@@ -105,6 +109,12 @@ struct sun4i_i2s_quirks {
 	const struct regmap_config	*sun4i_i2s_regmap;
 	unsigned int			mclk_offset;
 	unsigned int			bclk_offset;
+
+	/* Register fields for i2s */
+	struct reg_field		field_txchanmap;
+	struct reg_field		field_rxchanmap;
+	struct reg_field		field_txchansel;
+	struct reg_field		field_rxchansel;
 };
 
 struct sun4i_i2s {
@@ -118,6 +128,12 @@ struct sun4i_i2s {
 	struct snd_dmaengine_dai_dma_data	capture_dma_data;
 	struct snd_dmaengine_dai_dma_data	playback_dma_data;
 
+	/* Register fields for i2s */
+	struct regmap_field	*field_txchanmap;
+	struct regmap_field	*field_rxchanmap;
+	struct regmap_field	*field_txchansel;
+	struct regmap_field	*field_rxchansel;
+
 	const struct sun4i_i2s_quirks	*variant;
 };
 
@@ -268,6 +284,17 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (params_channels(params) != 2)
 		return -EINVAL;
 
+	/* Map the channels for playback and capture */
+	regmap_field_write(i2s->field_txchanmap, 0x76543210);
+	regmap_field_write(i2s->field_rxchanmap, 0x00003210);
+
+	/* Configure the channels */
+	regmap_field_write(i2s->field_txchansel,
+			   SUN4I_I2S_CHAN_SEL(params_channels(params)));
+
+	regmap_field_write(i2s->field_rxchansel,
+			   SUN4I_I2S_CHAN_SEL(params_channels(params)));
+
 	switch (params_physical_width(params)) {
 	case 16:
 		width = DMA_SLAVE_BUSWIDTH_2_BYTES;
@@ -490,13 +517,6 @@ static int sun4i_i2s_startup(struct snd_pcm_substream *substream,
 			   SUN4I_I2S_CTRL_SDO_EN_MASK,
 			   SUN4I_I2S_CTRL_SDO_EN(0));
 
-	/* Enable the first two channels */
-	regmap_write(i2s->regmap, SUN4I_I2S_TX_CHAN_SEL_REG,
-		     SUN4I_I2S_TX_CHAN_SEL(2));
-
-	/* Map them to the two first samples coming in */
-	regmap_write(i2s->regmap, SUN4I_I2S_TX_CHAN_MAP_REG,
-		     SUN4I_I2S_TX_CHAN_MAP(0, 0) | SUN4I_I2S_TX_CHAN_MAP(1, 1));
 
 	return clk_prepare_enable(i2s->mod_clk);
 }
@@ -681,14 +701,49 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.has_reset		= false,
 	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
+	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
+	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
+	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
 };
 
 static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
+	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
+	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
+	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
 };
 
+static int sun4i_i2s_init_regmap_fields(struct device *dev,
+					struct sun4i_i2s *i2s)
+{
+	i2s->field_txchanmap =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_txchanmap);
+	if (IS_ERR(i2s->field_txchanmap))
+		return PTR_ERR(i2s->field_txchanmap);
+
+	i2s->field_rxchanmap =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_rxchanmap);
+	if (IS_ERR(i2s->field_rxchanmap))
+		return PTR_ERR(i2s->field_rxchanmap);
+
+	i2s->field_txchansel =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_txchansel);
+	if (IS_ERR(i2s->field_txchansel))
+		return PTR_ERR(i2s->field_txchansel);
+
+	i2s->field_rxchansel =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_rxchansel);
+	return PTR_ERR_OR_ZERO(i2s->field_rxchansel);
+}
+
 static int sun4i_i2s_probe(struct platform_device *pdev)
 {
 	struct sun4i_i2s *i2s;
@@ -782,6 +837,12 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 		goto err_suspend;
 	}
 
+	ret = sun4i_i2s_init_regmap_fields(&pdev->dev, i2s);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not initialise regmap fields\n");
+		goto err_suspend;
+	}
+
 	return 0;
 
 err_suspend:
-- 
cgit v1.2.3


From 771647159125e93547f15daa14131a30a26f9b60 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:33 +0200
Subject: ASoC: sun4i-i2s: Add regfields for word size select and sample
 resolution

On newer SoCs the location of the slot width select and sample
resolution are different and also there is a bigger range of
support.

For the current supported rates then an offset is required.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 31 ++++++++++++++++++++++++++++---
 1 file changed, 28 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 87feb5a14cff..563de785d639 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -98,6 +98,9 @@
  * @sun4i_i2s_regmap: regmap config to use.
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
  * @bclk_offset: Value by which bclkdiv needs to be adjusted.
+ * @fmt_offset: Value by which wss and sr needs to be adjusted.
+ * @field_fmt_wss: regmap field to set word select size.
+ * @field_fmt_sr: regmap field to set sample resolution.
  * @field_txchanmap: location of the tx channel mapping register.
  * @field_rxchanmap: location of the rx channel mapping register.
  * @field_txchansel: location of the tx channel select bit fields.
@@ -109,8 +112,11 @@ struct sun4i_i2s_quirks {
 	const struct regmap_config	*sun4i_i2s_regmap;
 	unsigned int			mclk_offset;
 	unsigned int			bclk_offset;
+	unsigned int			fmt_offset;
 
 	/* Register fields for i2s */
+	struct reg_field		field_fmt_wss;
+	struct reg_field		field_fmt_sr;
 	struct reg_field		field_txchanmap;
 	struct reg_field		field_rxchanmap;
 	struct reg_field		field_txchansel;
@@ -129,6 +135,8 @@ struct sun4i_i2s {
 	struct snd_dmaengine_dai_dma_data	playback_dma_data;
 
 	/* Register fields for i2s */
+	struct regmap_field	*field_fmt_wss;
+	struct regmap_field	*field_fmt_sr;
 	struct regmap_field	*field_txchanmap;
 	struct regmap_field	*field_rxchanmap;
 	struct regmap_field	*field_txchansel;
@@ -314,9 +322,10 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
-			   SUN4I_I2S_FMT0_WSS_MASK | SUN4I_I2S_FMT0_SR_MASK,
-			   SUN4I_I2S_FMT0_WSS(wss) | SUN4I_I2S_FMT0_SR(sr));
+	regmap_field_write(i2s->field_fmt_wss,
+			   wss + i2s->variant->fmt_offset);
+	regmap_field_write(i2s->field_fmt_sr,
+			   sr + i2s->variant->fmt_offset);
 
 	return sun4i_i2s_set_clk_rate(i2s, params_rate(params),
 				      params_width(params));
@@ -701,6 +710,8 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.has_reset		= false,
 	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
@@ -711,6 +722,8 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
@@ -720,6 +733,18 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
 					struct sun4i_i2s *i2s)
 {
+	i2s->field_fmt_wss =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_fmt_wss);
+	if (IS_ERR(i2s->field_fmt_wss))
+		return PTR_ERR(i2s->field_fmt_wss);
+
+	i2s->field_fmt_sr =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_fmt_sr);
+	if (IS_ERR(i2s->field_fmt_sr))
+		return PTR_ERR(i2s->field_fmt_sr);
+
 	i2s->field_txchanmap =
 			devm_regmap_field_alloc(dev, i2s->regmap,
 						i2s->variant->field_txchanmap);
-- 
cgit v1.2.3


From 296935213feb7bcb72ab8750a8cacb66baeb0eb3 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:34 +0200
Subject: ASoC: sun4i-i2s: bclk and lrclk polarity tidyup

On newer SoCs the bit fields for the blck and lrclk polarity are in
a different locations. Use regmap fields to set the polarity bits
as intended.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 45 ++++++++++++++++++++++++++++++++-------------
 1 file changed, 32 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 563de785d639..8c7ad5bfe821 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -50,6 +50,8 @@
 #define SUN4I_I2S_FMT0_FMT_RIGHT_J			(2 << 0)
 #define SUN4I_I2S_FMT0_FMT_LEFT_J			(1 << 0)
 #define SUN4I_I2S_FMT0_FMT_I2S				(0 << 0)
+#define SUN4I_I2S_FMT0_POLARITY_INVERTED		(1)
+#define SUN4I_I2S_FMT0_POLARITY_NORMAL			(0)
 
 #define SUN4I_I2S_FMT1_REG		0x08
 #define SUN4I_I2S_FIFO_TX_REG		0x0c
@@ -101,6 +103,8 @@
  * @fmt_offset: Value by which wss and sr needs to be adjusted.
  * @field_fmt_wss: regmap field to set word select size.
  * @field_fmt_sr: regmap field to set sample resolution.
+ * @field_fmt_bclk: regmap field to set clk polarity.
+ * @field_fmt_lrclk: regmap field to set frame polarity.
  * @field_txchanmap: location of the tx channel mapping register.
  * @field_rxchanmap: location of the rx channel mapping register.
  * @field_txchansel: location of the tx channel select bit fields.
@@ -117,6 +121,8 @@ struct sun4i_i2s_quirks {
 	/* Register fields for i2s */
 	struct reg_field		field_fmt_wss;
 	struct reg_field		field_fmt_sr;
+	struct reg_field		field_fmt_bclk;
+	struct reg_field		field_fmt_lrclk;
 	struct reg_field		field_txchanmap;
 	struct reg_field		field_rxchanmap;
 	struct reg_field		field_txchansel;
@@ -137,6 +143,8 @@ struct sun4i_i2s {
 	/* Register fields for i2s */
 	struct regmap_field	*field_fmt_wss;
 	struct regmap_field	*field_fmt_sr;
+	struct regmap_field	*field_fmt_bclk;
+	struct regmap_field	*field_fmt_lrclk;
 	struct regmap_field	*field_txchanmap;
 	struct regmap_field	*field_rxchanmap;
 	struct regmap_field	*field_txchansel;
@@ -335,6 +343,8 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	u32 val;
+	u32 bclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
+	u32 lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
 
 	/* DAI Mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -359,32 +369,25 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_IB_IF:
 		/* Invert both clocks */
-		val = SUN4I_I2S_FMT0_BCLK_POLARITY_INVERTED |
-			SUN4I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
+		bclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
+		lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
 		break;
 	case SND_SOC_DAIFMT_IB_NF:
 		/* Invert bit clock */
-		val = SUN4I_I2S_FMT0_BCLK_POLARITY_INVERTED |
-			SUN4I_I2S_FMT0_LRCLK_POLARITY_NORMAL;
+		bclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
 		break;
 	case SND_SOC_DAIFMT_NB_IF:
 		/* Invert frame clock */
-		val = SUN4I_I2S_FMT0_LRCLK_POLARITY_INVERTED |
-			SUN4I_I2S_FMT0_BCLK_POLARITY_NORMAL;
+		lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
 		break;
 	case SND_SOC_DAIFMT_NB_NF:
-		/* Nothing to do for both normal cases */
-		val = SUN4I_I2S_FMT0_BCLK_POLARITY_NORMAL |
-			SUN4I_I2S_FMT0_LRCLK_POLARITY_NORMAL;
 		break;
 	default:
 		return -EINVAL;
 	}
 
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
-			   SUN4I_I2S_FMT0_BCLK_POLARITY_MASK |
-			   SUN4I_I2S_FMT0_LRCLK_POLARITY_MASK,
-			   val);
+	regmap_field_write(i2s->field_fmt_bclk, bclk_polarity);
+	regmap_field_write(i2s->field_fmt_lrclk, lrclk_polarity);
 
 	/* DAI clock master masks */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -712,6 +715,8 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
+	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
@@ -724,6 +729,8 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
+	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
@@ -745,6 +752,18 @@ static int sun4i_i2s_init_regmap_fields(struct device *dev,
 	if (IS_ERR(i2s->field_fmt_sr))
 		return PTR_ERR(i2s->field_fmt_sr);
 
+	i2s->field_fmt_bclk =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_fmt_bclk);
+	if (IS_ERR(i2s->field_fmt_bclk))
+		return PTR_ERR(i2s->field_fmt_bclk);
+
+	i2s->field_fmt_lrclk =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_fmt_lrclk);
+	if (IS_ERR(i2s->field_fmt_lrclk))
+		return PTR_ERR(i2s->field_fmt_lrclk);
+
 	i2s->field_txchanmap =
 			devm_regmap_field_alloc(dev, i2s->regmap,
 						i2s->variant->field_txchanmap);
-- 
cgit v1.2.3


From 5f93b0639634929b8349f9f9d83e1d881c1c7d4a Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:35 +0200
Subject: ASoC: sun4i-i2s: Add mclk enable regmap field

The location of the mclk output enable bit is different on newer
SoCs. Use a regmap field to enable it.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 16 ++++++++++++++--
 1 file changed, 14 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 8c7ad5bfe821..761b7c0f6a32 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -101,6 +101,7 @@
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
  * @bclk_offset: Value by which bclkdiv needs to be adjusted.
  * @fmt_offset: Value by which wss and sr needs to be adjusted.
+ * @field_clkdiv_mclk_en: regmap field to enable mclk output.
  * @field_fmt_wss: regmap field to set word select size.
  * @field_fmt_sr: regmap field to set sample resolution.
  * @field_fmt_bclk: regmap field to set clk polarity.
@@ -119,6 +120,7 @@ struct sun4i_i2s_quirks {
 	unsigned int			fmt_offset;
 
 	/* Register fields for i2s */
+	struct reg_field		field_clkdiv_mclk_en;
 	struct reg_field		field_fmt_wss;
 	struct reg_field		field_fmt_sr;
 	struct reg_field		field_fmt_bclk;
@@ -141,6 +143,7 @@ struct sun4i_i2s {
 	struct snd_dmaengine_dai_dma_data	playback_dma_data;
 
 	/* Register fields for i2s */
+	struct regmap_field	*field_clkdiv_mclk_en;
 	struct regmap_field	*field_fmt_wss;
 	struct regmap_field	*field_fmt_sr;
 	struct regmap_field	*field_fmt_bclk;
@@ -283,8 +286,9 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s,
 
 	regmap_write(i2s->regmap, SUN4I_I2S_CLK_DIV_REG,
 		     SUN4I_I2S_CLK_DIV_BCLK(bclk_div) |
-		     SUN4I_I2S_CLK_DIV_MCLK(mclk_div) |
-		     SUN4I_I2S_CLK_DIV_MCLK_EN);
+		     SUN4I_I2S_CLK_DIV_MCLK(mclk_div));
+
+	regmap_field_write(i2s->field_clkdiv_mclk_en, 1);
 
 	return 0;
 }
@@ -713,6 +717,7 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.has_reset		= false,
 	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
@@ -727,6 +732,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN4I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
@@ -740,6 +746,12 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
 					struct sun4i_i2s *i2s)
 {
+	i2s->field_clkdiv_mclk_en =
+		devm_regmap_field_alloc(dev, i2s->regmap,
+					i2s->variant->field_clkdiv_mclk_en);
+	if (IS_ERR(i2s->field_clkdiv_mclk_en))
+		return PTR_ERR(i2s->field_clkdiv_mclk_en);
+
 	i2s->field_fmt_wss =
 			devm_regmap_field_alloc(dev, i2s->regmap,
 						i2s->variant->field_fmt_wss);
-- 
cgit v1.2.3


From dfd2293c8626f1282e6a6e30f591c0a83146f3d4 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:36 +0200
Subject: ASoC: sun4i-i2s: Add regmap field to set DAI format

On the newer SoCs the bits to configure the operational mode are
located in a different register. Add a regmap field so that this
location can be configured.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 15 ++++++++++++---
 1 file changed, 12 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 761b7c0f6a32..d9910f6617ad 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -106,6 +106,7 @@
  * @field_fmt_sr: regmap field to set sample resolution.
  * @field_fmt_bclk: regmap field to set clk polarity.
  * @field_fmt_lrclk: regmap field to set frame polarity.
+ * @field_fmt_mode: regmap field to set the operational mode.
  * @field_txchanmap: location of the tx channel mapping register.
  * @field_rxchanmap: location of the rx channel mapping register.
  * @field_txchansel: location of the tx channel select bit fields.
@@ -125,6 +126,7 @@ struct sun4i_i2s_quirks {
 	struct reg_field		field_fmt_sr;
 	struct reg_field		field_fmt_bclk;
 	struct reg_field		field_fmt_lrclk;
+	struct reg_field		field_fmt_mode;
 	struct reg_field		field_txchanmap;
 	struct reg_field		field_rxchanmap;
 	struct reg_field		field_txchansel;
@@ -148,6 +150,7 @@ struct sun4i_i2s {
 	struct regmap_field	*field_fmt_sr;
 	struct regmap_field	*field_fmt_bclk;
 	struct regmap_field	*field_fmt_lrclk;
+	struct regmap_field	*field_fmt_mode;
 	struct regmap_field	*field_txchanmap;
 	struct regmap_field	*field_rxchanmap;
 	struct regmap_field	*field_txchansel;
@@ -365,9 +368,7 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
-			   SUN4I_I2S_FMT0_FMT_MASK,
-			   val);
+	regmap_field_write(i2s->field_fmt_mode, val);
 
 	/* DAI clock polarity */
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -722,6 +723,7 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
@@ -737,6 +739,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
@@ -776,6 +779,12 @@ static int sun4i_i2s_init_regmap_fields(struct device *dev,
 	if (IS_ERR(i2s->field_fmt_lrclk))
 		return PTR_ERR(i2s->field_fmt_lrclk);
 
+	i2s->field_fmt_mode =
+			devm_regmap_field_alloc(dev, i2s->regmap,
+						i2s->variant->field_fmt_mode);
+	if (IS_ERR(i2s->field_fmt_mode))
+		return PTR_ERR(i2s->field_fmt_mode);
+
 	i2s->field_txchanmap =
 			devm_regmap_field_alloc(dev, i2s->regmap,
 						i2s->variant->field_txchanmap);
-- 
cgit v1.2.3


From d03d2737e2d20c6ffb242154a69e1a9313adb632 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:37 +0200
Subject: ASoC: sun4i-i2s: Check for slave select bit

The newer SoCs do not have this setting. Instead they set the pin
direction. Add a check to see if the bit is valid and if so set
it accordingly.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 37 +++++++++++++++++++++----------------
 1 file changed, 21 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index d9910f6617ad..573acca24a2c 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -96,6 +96,7 @@
  * struct sun4i_i2s_quirks - Differences between SoC variants.
  *
  * @has_reset: SoC needs reset deasserted.
+ * @has_slave_select_bit: SoC has a bit to enable slave mode.
  * @reg_offset_txdata: offset of the tx fifo.
  * @sun4i_i2s_regmap: regmap config to use.
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
@@ -114,6 +115,7 @@
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
+	bool				has_slave_select_bit;
 	unsigned int			reg_offset_txdata;	/* TX FIFO */
 	const struct regmap_config	*sun4i_i2s_regmap;
 	unsigned int			mclk_offset;
@@ -394,24 +396,25 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	regmap_field_write(i2s->field_fmt_bclk, bclk_polarity);
 	regmap_field_write(i2s->field_fmt_lrclk, lrclk_polarity);
 
-	/* DAI clock master masks */
-	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBS_CFS:
-		/* BCLK and LRCLK master */
-		val = SUN4I_I2S_CTRL_MODE_MASTER;
-		break;
-	case SND_SOC_DAIFMT_CBM_CFM:
-		/* BCLK and LRCLK slave */
-		val = SUN4I_I2S_CTRL_MODE_SLAVE;
-		break;
-	default:
-		return -EINVAL;
+	if (i2s->variant->has_slave_select_bit) {
+		/* DAI clock master masks */
+		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBS_CFS:
+			/* BCLK and LRCLK master */
+			val = SUN4I_I2S_CTRL_MODE_MASTER;
+			break;
+		case SND_SOC_DAIFMT_CBM_CFM:
+			/* BCLK and LRCLK slave */
+			val = SUN4I_I2S_CTRL_MODE_SLAVE;
+			break;
+		default:
+			return -EINVAL;
+		}
+		regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+				   SUN4I_I2S_CTRL_MODE_MASK,
+				   val);
 	}
 
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_MODE_MASK,
-			   val);
-
 	/* Set significant bits in our FIFOs */
 	regmap_update_bits(i2s->regmap, SUN4I_I2S_FIFO_CTRL_REG,
 			   SUN4I_I2S_FIFO_CTRL_TX_MODE_MASK |
@@ -723,6 +726,7 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.has_slave_select_bit	= true,
 	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
@@ -739,6 +743,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.has_slave_select_bit	= true,
 	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
-- 
cgit v1.2.3


From 043b8daa578f30c2ba685ad13b121fd138b95750 Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:38 +0200
Subject: ASoC: sun4i-i2s: Update global enable with bitmask

The default value of the config register is different on newer
SoCs and therefore enabling/disabling with a register write
will clear bits used to set the direction of the clock and frame
pins.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 573acca24a2c..19d50ca10d1f 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -529,8 +529,8 @@ static int sun4i_i2s_startup(struct snd_pcm_substream *substream,
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 
 	/* Enable the whole hardware block */
-	regmap_write(i2s->regmap, SUN4I_I2S_CTRL_REG,
-		     SUN4I_I2S_CTRL_GL_EN);
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
 
 	/* Enable the first output line */
 	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
@@ -553,7 +553,8 @@ static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream,
 			   SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
 
 	/* Disable the whole hardware block */
-	regmap_write(i2s->regmap, SUN4I_I2S_CTRL_REG, 0);
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_GL_EN, 0);
 }
 
 static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
-- 
cgit v1.2.3


From 7d2993811a1e07cd128847f07f63aae66a78d14b Mon Sep 17 00:00:00 2001
From: Marcus Cooper
Date: Sat, 19 Aug 2017 14:48:39 +0200
Subject: ASoC: sun4i-i2s: Add support for H3

The sun8i-h3 introduces a lot of changes to the i2s block such
as different register locations, extended clock division and
more operational modes. As we have to consider the earlier
implementation then these changes need to be isolated.

None of the new functionality has been implemented yet, the
driver has just been expanded to allow it work on the H3 SoC.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/sun4i-i2s.txt        |   2 +
 sound/soc/sunxi/sun4i-i2s.c                        | 176 ++++++++++++++++++++-
 2 files changed, 176 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
index ee21da865771..fc5da6080759 100644
--- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt
@@ -8,6 +8,7 @@ Required properties:
 - compatible: should be one of the following:
    - "allwinner,sun4i-a10-i2s"
    - "allwinner,sun6i-a31-i2s"
+   - "allwinner,sun8i-h3-i2s"
 - reg: physical base address of the controller and length of memory mapped
   region.
 - interrupts: should contain the I2S interrupt.
@@ -22,6 +23,7 @@ Required properties:
 
 Required properties for the following compatibles:
 	- "allwinner,sun6i-a31-i2s"
+	- "allwinner,sun8i-h3-i2s"
 - resets: phandle to the reset line for this codec
 
 Example:
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 19d50ca10d1f..04f92583a969 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -92,11 +92,41 @@
 #define SUN4I_I2S_RX_CHAN_SEL_REG	0x38
 #define SUN4I_I2S_RX_CHAN_MAP_REG	0x3c
 
+/* Defines required for sun8i-h3 support */
+#define SUN8I_I2S_CTRL_BCLK_OUT			BIT(18)
+#define SUN8I_I2S_CTRL_LRCK_OUT			BIT(17)
+
+#define SUN8I_I2S_FMT0_LRCK_PERIOD_MASK		GENMASK(17, 8)
+#define SUN8I_I2S_FMT0_LRCK_PERIOD(period)	((period - 1) << 8)
+
+#define SUN8I_I2S_INT_STA_REG		0x0c
+#define SUN8I_I2S_FIFO_TX_REG		0x20
+
+#define SUN8I_I2S_CHAN_CFG_REG		0x30
+#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK	GENMASK(6, 4)
+#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan)	(chan - 1)
+#define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK	GENMASK(2, 0)
+#define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan)	(chan - 1)
+
+#define SUN8I_I2S_TX_CHAN_MAP_REG	0x44
+#define SUN8I_I2S_TX_CHAN_SEL_REG	0x34
+#define SUN8I_I2S_TX_CHAN_OFFSET_MASK		GENMASK(13, 11)
+#define SUN8I_I2S_TX_CHAN_OFFSET(offset)	(offset << 12)
+#define SUN8I_I2S_TX_CHAN_EN_MASK		GENMASK(11, 4)
+#define SUN8I_I2S_TX_CHAN_EN(num_chan)		(((1 << num_chan) - 1) << 4)
+
+#define SUN8I_I2S_RX_CHAN_SEL_REG	0x54
+#define SUN8I_I2S_RX_CHAN_MAP_REG	0x58
+
 /**
  * struct sun4i_i2s_quirks - Differences between SoC variants.
  *
  * @has_reset: SoC needs reset deasserted.
  * @has_slave_select_bit: SoC has a bit to enable slave mode.
+ * @has_fmt_set_lrck_period: SoC requires lrclk period to be set.
+ * @has_chcfg: tx and rx slot number need to be set.
+ * @has_chsel_tx_chen: SoC requires that the tx channels are enabled.
+ * @has_chsel_offset: SoC uses offset for selecting dai operational mode.
  * @reg_offset_txdata: offset of the tx fifo.
  * @sun4i_i2s_regmap: regmap config to use.
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
@@ -116,6 +146,10 @@
 struct sun4i_i2s_quirks {
 	bool				has_reset;
 	bool				has_slave_select_bit;
+	bool				has_fmt_set_lrck_period;
+	bool				has_chcfg;
+	bool				has_chsel_tx_chen;
+	bool				has_chsel_offset;
 	unsigned int			reg_offset_txdata;	/* TX FIFO */
 	const struct regmap_config	*sun4i_i2s_regmap;
 	unsigned int			mclk_offset;
@@ -173,6 +207,7 @@ static const struct sun4i_i2s_clk_div sun4i_i2s_bclk_div[] = {
 	{ .div = 8, .val = 3 },
 	{ .div = 12, .val = 4 },
 	{ .div = 16, .val = 5 },
+	/* TODO - extend divide ratio supported by newer SoCs */
 };
 
 static const struct sun4i_i2s_clk_div sun4i_i2s_mclk_div[] = {
@@ -184,6 +219,7 @@ static const struct sun4i_i2s_clk_div sun4i_i2s_mclk_div[] = {
 	{ .div = 12, .val = 5 },
 	{ .div = 16, .val = 6 },
 	{ .div = 24, .val = 7 },
+	/* TODO - extend divide ratio supported by newer SoCs */
 };
 
 static int sun4i_i2s_get_bclk_div(struct sun4i_i2s *i2s,
@@ -295,6 +331,12 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s,
 
 	regmap_field_write(i2s->field_clkdiv_mclk_en, 1);
 
+	/* Set sync period */
+	if (i2s->variant->has_fmt_set_lrck_period)
+		regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
+				   SUN8I_I2S_FMT0_LRCK_PERIOD_MASK,
+				   SUN8I_I2S_FMT0_LRCK_PERIOD(32));
+
 	return 0;
 }
 
@@ -303,12 +345,22 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 			       struct snd_soc_dai *dai)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	int sr, wss;
+	int sr, wss, channels;
 	u32 width;
 
-	if (params_channels(params) != 2)
+	channels = params_channels(params);
+	if (channels != 2)
 		return -EINVAL;
 
+	if (i2s->variant->has_chcfg) {
+		regmap_update_bits(i2s->regmap, SUN8I_I2S_CHAN_CFG_REG,
+				   SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK,
+				   SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(channels));
+		regmap_update_bits(i2s->regmap, SUN8I_I2S_CHAN_CFG_REG,
+				   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK,
+				   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(channels));
+	}
+
 	/* Map the channels for playback and capture */
 	regmap_field_write(i2s->field_txchanmap, 0x76543210);
 	regmap_field_write(i2s->field_rxchanmap, 0x00003210);
@@ -320,6 +372,11 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 	regmap_field_write(i2s->field_rxchansel,
 			   SUN4I_I2S_CHAN_SEL(params_channels(params)));
 
+	if (i2s->variant->has_chsel_tx_chen)
+		regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
+				   SUN8I_I2S_TX_CHAN_EN_MASK,
+				   SUN8I_I2S_TX_CHAN_EN(channels));
+
 	switch (params_physical_width(params)) {
 	case 16:
 		width = DMA_SLAVE_BUSWIDTH_2_BYTES;
@@ -352,6 +409,7 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	u32 val;
+	u32 offset = 0;
 	u32 bclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
 	u32 lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
 
@@ -359,6 +417,7 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		val = SUN4I_I2S_FMT0_FMT_I2S;
+		offset = 1;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
 		val = SUN4I_I2S_FMT0_FMT_LEFT_J;
@@ -370,6 +429,21 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
+	if (i2s->variant->has_chsel_offset) {
+		/*
+		 * offset being set indicates that we're connected to an i2s
+		 * device, however offset is only used on the sun8i block and
+		 * i2s shares the same setting with the LJ format. Increment
+		 * val so that the bit to value to write is correct.
+		 */
+		if (offset > 0)
+			val++;
+		/* blck offset determines whether i2s or LJ */
+		regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
+				   SUN8I_I2S_TX_CHAN_OFFSET_MASK,
+				   SUN8I_I2S_TX_CHAN_OFFSET(offset));
+	}
+
 	regmap_field_write(i2s->field_fmt_mode, val);
 
 	/* DAI clock polarity */
@@ -413,6 +487,29 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
 				   SUN4I_I2S_CTRL_MODE_MASK,
 				   val);
+	} else {
+		/*
+		 * The newer i2s block does not have a slave select bit,
+		 * instead the clk pins are configured as inputs.
+		 */
+		/* DAI clock master masks */
+		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBS_CFS:
+			/* BCLK and LRCLK master */
+			val = SUN8I_I2S_CTRL_BCLK_OUT |
+				SUN8I_I2S_CTRL_LRCK_OUT;
+			break;
+		case SND_SOC_DAIFMT_CBM_CFM:
+			/* BCLK and LRCLK slave */
+			val = 0;
+			break;
+		default:
+			return -EINVAL;
+		}
+		regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+				   SUN8I_I2S_CTRL_BCLK_OUT |
+				   SUN8I_I2S_CTRL_LRCK_OUT,
+				   val);
 	}
 
 	/* Set significant bits in our FIFOs */
@@ -653,6 +750,27 @@ static bool sun4i_i2s_volatile_reg(struct device *dev, unsigned int reg)
 	}
 }
 
+static bool sun8i_i2s_rd_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case SUN8I_I2S_FIFO_TX_REG:
+		return false;
+
+	default:
+		return true;
+	}
+}
+
+static bool sun8i_i2s_volatile_reg(struct device *dev, unsigned int reg)
+{
+	if (reg == SUN8I_I2S_INT_STA_REG)
+		return true;
+	if (reg == SUN8I_I2S_FIFO_TX_REG)
+		return false;
+
+	return sun4i_i2s_volatile_reg(dev, reg);
+}
+
 static const struct reg_default sun4i_i2s_reg_defaults[] = {
 	{ SUN4I_I2S_CTRL_REG, 0x00000000 },
 	{ SUN4I_I2S_FMT0_REG, 0x0000000c },
@@ -666,6 +784,20 @@ static const struct reg_default sun4i_i2s_reg_defaults[] = {
 	{ SUN4I_I2S_RX_CHAN_MAP_REG, 0x00003210 },
 };
 
+static const struct reg_default sun8i_i2s_reg_defaults[] = {
+	{ SUN4I_I2S_CTRL_REG, 0x00060000 },
+	{ SUN4I_I2S_FMT0_REG, 0x00000033 },
+	{ SUN4I_I2S_FMT1_REG, 0x00000030 },
+	{ SUN4I_I2S_FIFO_CTRL_REG, 0x000400f0 },
+	{ SUN4I_I2S_DMA_INT_CTRL_REG, 0x00000000 },
+	{ SUN4I_I2S_CLK_DIV_REG, 0x00000000 },
+	{ SUN8I_I2S_CHAN_CFG_REG, 0x00000000 },
+	{ SUN8I_I2S_TX_CHAN_SEL_REG, 0x00000000 },
+	{ SUN8I_I2S_TX_CHAN_MAP_REG, 0x00000000 },
+	{ SUN8I_I2S_RX_CHAN_SEL_REG, 0x00000000 },
+	{ SUN8I_I2S_RX_CHAN_MAP_REG, 0x00000000 },
+};
+
 static const struct regmap_config sun4i_i2s_regmap_config = {
 	.reg_bits	= 32,
 	.reg_stride	= 4,
@@ -680,6 +812,19 @@ static const struct regmap_config sun4i_i2s_regmap_config = {
 	.volatile_reg	= sun4i_i2s_volatile_reg,
 };
 
+static const struct regmap_config sun8i_i2s_regmap_config = {
+	.reg_bits	= 32,
+	.reg_stride	= 4,
+	.val_bits	= 32,
+	.max_register	= SUN8I_I2S_RX_CHAN_MAP_REG,
+	.cache_type	= REGCACHE_FLAT,
+	.reg_defaults	= sun8i_i2s_reg_defaults,
+	.num_reg_defaults	= ARRAY_SIZE(sun8i_i2s_reg_defaults),
+	.writeable_reg	= sun4i_i2s_wr_reg,
+	.readable_reg	= sun8i_i2s_rd_reg,
+	.volatile_reg	= sun8i_i2s_volatile_reg,
+};
+
 static int sun4i_i2s_runtime_resume(struct device *dev)
 {
 	struct sun4i_i2s *i2s = dev_get_drvdata(dev);
@@ -752,6 +897,29 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
 };
 
+static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
+	.has_reset		= true,
+	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
+	.sun4i_i2s_regmap	= &sun8i_i2s_regmap_config,
+	.mclk_offset		= 1,
+	.bclk_offset		= 2,
+	.fmt_offset		= 3,
+	.has_fmt_set_lrck_period = true,
+	.has_chcfg		= true,
+	.has_chsel_tx_chen	= true,
+	.has_chsel_offset	= true,
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
+	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 19, 19),
+	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_CTRL_REG, 4, 5),
+	.field_txchanmap	= REG_FIELD(SUN8I_I2S_TX_CHAN_MAP_REG, 0, 31),
+	.field_rxchanmap	= REG_FIELD(SUN8I_I2S_RX_CHAN_MAP_REG, 0, 31),
+	.field_txchansel	= REG_FIELD(SUN8I_I2S_TX_CHAN_SEL_REG, 0, 2),
+	.field_rxchansel	= REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2),
+};
+
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
 					struct sun4i_i2s *i2s)
 {
@@ -952,6 +1120,10 @@ static const struct of_device_id sun4i_i2s_match[] = {
 		.compatible = "allwinner,sun6i-a31-i2s",
 		.data = &sun6i_a31_i2s_quirks,
 	},
+	{
+		.compatible = "allwinner,sun8i-h3-i2s",
+		.data = &sun8i_h3_i2s_quirks,
+	},
 	{}
 };
 MODULE_DEVICE_TABLE(of, sun4i_i2s_match);
-- 
cgit v1.2.3


From e5d619e9b924c03b4e5e6be7c03f1346150104b9 Mon Sep 17 00:00:00 2001
From: Lucas Stach
Date: Fri, 18 Aug 2017 18:33:35 +0200
Subject: ASoC: fsl-asoc-card: don't print EPROBE_DEFER as error

Probe deferral is to be expected during normal operation, so avoid
printing an error when it is encountered.

Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl-asoc-card.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 9998aea23597..2db4d0c80d33 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -683,7 +683,7 @@ 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)
+	if (ret && ret != -EPROBE_DEFER)
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
 
 asrc_fail:
-- 
cgit v1.2.3


From b286401752c61de9c2335a1c9748243bb53649de Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 13:22:00 +0800
Subject: ASoC: mediatek: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to card->dev,then
the function platform_set_drvdata copies the value the variable card
to pdev->dev.driver_data, but when calling snd_soc_register_card,the
function dev_set_drvdata(card->dev, card) will also do the same copy
operation,so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 -
 sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 -
 sound/soc/mediatek/mt8173/mt8173-rt5650.c        | 1 -
 3 files changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
index 5e383eb456a4..99c15219dbc8 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
@@ -222,7 +222,6 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
 		mt8173_rt5650_rt5514_codecs[1].of_node;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index fed1f15a39c2..42de84ca8c84 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -279,7 +279,6 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 	}
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index a78470839b65..e69c141d8ed4 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -311,7 +311,6 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
-- 
cgit v1.2.3


From e4eabf75ad9a0d4afb4a35c8c4fac7e437d5da06 Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 14:02:17 +0800
Subject: ASoC: tegra: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to card->dev,then
the function platform_set_drvdata copies the value the variable card
to pdev->dev.driver_data, but when calling snd_soc_register_card,the
function dev_set_drvdata(card->dev, card) will also do the same copy
operation,so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra_alc5632.c  | 1 -
 sound/soc/tegra/tegra_max98090.c | 1 -
 sound/soc/tegra/tegra_rt5640.c   | 1 -
 sound/soc/tegra/tegra_rt5677.c   | 1 -
 sound/soc/tegra/tegra_sgtl5000.c | 1 -
 sound/soc/tegra/tegra_wm8753.c   | 1 -
 sound/soc/tegra/tegra_wm8903.c   | 1 -
 sound/soc/tegra/tegra_wm9712.c   | 1 -
 sound/soc/tegra/trimslice.c      | 1 -
 9 files changed, 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 0509902512cc..053fe4c5ed4c 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -173,7 +173,6 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, alc5632);
 
 	alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
index c34a54d6e812..41feee9beb12 100644
--- a/sound/soc/tegra/tegra_max98090.c
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -229,7 +229,6 @@ static int tegra_max98090_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
 	machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index 93a356802345..02f27fe14324 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -174,7 +174,6 @@ static int tegra_rt5640_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
 	machine->gpio_hp_det = of_get_named_gpio_flags(
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index ebf58d0e0f10..3dbf0e1350c1 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -222,7 +222,6 @@ static int tegra_rt5677_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
 	machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index 6dda01f69983..45a4aa9d2a47 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -124,7 +124,6 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
 	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index d0ab0026a4cd..23a810e3bacc 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -132,7 +132,6 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
 	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index dbfb49298ae8..d4a4a2be8b42 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -252,7 +252,6 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
 	machine->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios",
diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c
index c9cd22432627..864a3345972e 100644
--- a/sound/soc/tegra/tegra_wm9712.c
+++ b/sound/soc/tegra/tegra_wm9712.c
@@ -81,7 +81,6 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
 	machine->codec = platform_device_alloc("wm9712-codec", -1);
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index c9dcad9bb931..99bcdd979eb2 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -127,7 +127,6 @@ static int tegra_snd_trimslice_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, trimslice);
 
 	trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle(np,
-- 
cgit v1.2.3


From c996b52df44b83d227d7664d2838ed1c6739c850 Mon Sep 17 00:00:00 2001
From: Donglin Peng
Date: Sun, 20 Aug 2017 14:01:10 +0800
Subject: ASoC: ux500: Remove unnecessary function call

First of all,the address of pdev->dev is assigned to mop500_card.dev,
then the function platform_set_drvdata copies the value the variable
card to pdev->dev.driver_data,but when calling snd_soc_register_card,
the function dev_set_drvdata(card->dev, card) will also do the same
copy operation,so i think that the former copy operation can be removed.

Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ux500/mop500.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index b50f68a439ce..9298580e4c2a 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -111,7 +111,6 @@ static int mop500_probe(struct platform_device *pdev)
 
 	dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n",
 		__func__, mop500_card.name);
-	platform_set_drvdata(pdev, &mop500_card);
 
 	snd_soc_card_set_drvdata(&mop500_card, NULL);
 
-- 
cgit v1.2.3


From e269998d588f8ad96eaf86916e23b8ee3d2b9f1b Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Thu, 17 Aug 2017 10:02:09 +0200
Subject: ASoC: codecs: msm8916-wcd-analog: get micbias voltage from dt

This patch adds bindings in DT to provide required micbias voltage which
could be specific to board. With this new binding, now the mic bias
voltage is left at hardware default value if the device tree does not
specify any mic bias voltage value. Correct micbias value is required
for mbhc buttons to work.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../bindings/sound/qcom,msm8916-wcd-analog.txt     |  1 +
 sound/soc/codecs/msm8916-wcd-analog.c              | 27 +++++++++++++++++-----
 2 files changed, 22 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
index ccb401cfef9d..05b67a1d4851 100644
--- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
@@ -33,6 +33,7 @@ Required properties
  - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node.
 
 Optional Properties:
+- qcom,micbias-lvl:  Voltage (mV) for Mic Bias
 - qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor
 			 connected.
 - qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index a0de3e7204d5..8633524b1961 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -93,8 +93,12 @@
 #define MICB_1_EN_TX3_GND_SEL_TX_GND	0
 
 #define CDC_A_MICB_1_VAL		(0xf141)
+#define MICB_MIN_VAL 1600
+#define MICB_STEP_SIZE 50
+#define MICB_VOLTAGE_REGVAL(v)		((v - MICB_MIN_VAL)/MICB_STEP_SIZE)
 #define MICB_1_VAL_MICB_OUT_VAL_MASK	GENMASK(7, 3)
 #define MICB_1_VAL_MICB_OUT_VAL_V2P70V	((0x16)  << 3)
+#define MICB_1_VAL_MICB_OUT_VAL_V1P80V	((0x4)  << 3)
 #define CDC_A_MICB_1_CTL		(0xf142)
 
 #define MICB_1_CTL_CFILT_REF_SEL_MASK		BIT(1)
@@ -225,6 +229,7 @@ struct pm8916_wcd_analog_priv {
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 	unsigned int micbias1_cap_mode;
 	unsigned int micbias2_cap_mode;
+	unsigned int micbias_mv;
 };
 
 static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
@@ -265,18 +270,25 @@ static const struct snd_kcontrol_new pm8916_wcd_analog_snd_controls[] = {
 
 static void pm8916_wcd_analog_micbias_enable(struct snd_soc_codec *codec)
 {
+	struct pm8916_wcd_analog_priv *wcd = snd_soc_codec_get_drvdata(codec);
+
 	snd_soc_update_bits(codec, CDC_A_MICB_1_CTL,
 			    MICB_1_CTL_EXT_PRECHARG_EN_MASK |
 			    MICB_1_CTL_INT_PRECHARG_BYP_MASK,
 			    MICB_1_CTL_INT_PRECHARG_BYP_EXT_PRECHRG_SEL
 			    | MICB_1_CTL_EXT_PRECHARG_EN_ENABLE);
 
-	snd_soc_write(codec, CDC_A_MICB_1_VAL, MICB_1_VAL_MICB_OUT_VAL_V2P70V);
-	/*
-	 * Special headset needs MICBIAS as 2.7V so wait for
-	 * 50 msec for the MICBIAS to reach 2.7 volts.
-	 */
-	msleep(50);
+	if (wcd->micbias_mv) {
+		snd_soc_write(codec, CDC_A_MICB_1_VAL,
+			      MICB_VOLTAGE_REGVAL(wcd->micbias_mv));
+		/*
+		 * Special headset needs MICBIAS as 2.7V so wait for
+		 * 50 msec for the MICBIAS to reach 2.7 volts.
+		 */
+		if (wcd->micbias_mv >= 2700)
+			msleep(50);
+	}
+
 	snd_soc_update_bits(codec, CDC_A_MICB_1_CTL,
 			    MICB_1_CTL_EXT_PRECHARG_EN_MASK |
 			    MICB_1_CTL_INT_PRECHARG_BYP_MASK, 0);
@@ -795,6 +807,9 @@ static int pm8916_wcd_analog_parse_dt(struct device *dev,
 	else
 		priv->micbias2_cap_mode = MICB_1_EN_NO_EXT_BYP_CAP;
 
+	of_property_read_u32(dev->of_node, "qcom,micbias-lvl",
+			     &priv->micbias_mv);
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From de66b3455023e6f78fdf55a387c604c6b0114869 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Thu, 17 Aug 2017 10:02:10 +0200
Subject: ASoC: codecs: msm8916-wcd-analog: add MBHC support

MBHC (MultiButton Headset Control) support is available in pm8921 in two
blocks, one to detect mechanical headset insertion and removal and other
block to support headset type detection and 5 button detection and othe
features like impedance calculation.

This patch adds support to:
1> Support to NC and NO type of headset Jacks.
2> Mechanical insertion and detection of headset jack.
3> Detect a 3 pole Headphone and a 4 pole Headset.
4> Detect 5 buttons.

Tested it on DB410c with Audio Mezz board with 4 pole and 3 pole
headset/headphones.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Damien Riegel <damien.riegel@savoirfairelinux.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../bindings/sound/qcom,msm8916-wcd-analog.txt     |  17 +-
 sound/soc/codecs/msm8916-wcd-analog.c              | 373 ++++++++++++++++++++-
 2 files changed, 388 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
index 05b67a1d4851..551ecab67efe 100644
--- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt
@@ -31,9 +31,22 @@ Required properties
  - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node.
  - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node.
  - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node.
-
 Optional Properties:
+ - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons
+			     detection on headset when the mbhc is powered up
+			     by internal current source, this is a low power.
+ - qcom,mbhc-vthreshold-high: Array of 5 thresold voltages in mV for 5 buttons
+			      detection on headset when mbhc is powered up
+			       from micbias.
 - qcom,micbias-lvl:  Voltage (mV) for Mic Bias
+- qcom,hphl-jack-type-normally-open: boolean, present if hphl pin on jack is a
+				     NO (Normally Open). If not specified, then
+				     its assumed that hphl pin on jack is NC
+				     (Normally Closed).
+- qcom,gnd-jack-type-normally-open: boolean, present if gnd pin on jack is
+				    NO (Normally Open). If not specified, then
+				    its assumed that gnd pin on jack is NC
+				    (Normally Closed).
 - qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor
 			 connected.
 - qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor
@@ -49,6 +62,8 @@ spmi_bus {
 		reg-names = "pmic-codec-core";
 		clocks = <&gcc GCC_CODEC_DIGCODEC_CLK>;
 		clock-names = "mclk";
+		qcom,mbhc-vthreshold-low = <75 150 237 450 500>;
+		qcom,mbhc-vthreshold-high = <75 150 237 450 500>;
 		interrupt-parent = <&spmi_bus>;
 		interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>,
 			     <0x1 0xf0 0x1 IRQ_TYPE_NONE>,
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index 8633524b1961..f834a639b350 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -12,9 +12,16 @@
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/tlv.h>
+#include <sound/jack.h>
 
 #define CDC_D_REVISION1			(0xf000)
 #define CDC_D_PERPH_SUBTYPE		(0xf005)
+#define CDC_D_INT_EN_SET		(0x015)
+#define CDC_D_INT_EN_CLR		(0x016)
+#define MBHC_SWITCH_INT			BIT(7)
+#define MBHC_MIC_ELECTRICAL_INS_REM_DET	BIT(6)
+#define MBHC_BUTTON_PRESS_DET		BIT(5)
+#define MBHC_BUTTON_RELEASE_DET		BIT(4)
 #define CDC_D_CDC_RST_CTL		(0xf046)
 #define RST_CTL_DIG_SW_RST_N_MASK	BIT(7)
 #define RST_CTL_DIG_SW_RST_N_RESET	0
@@ -37,6 +44,8 @@
 #define DIG_CLK_CTL_RXD1_CLK_EN		BIT(0)
 #define DIG_CLK_CTL_RXD2_CLK_EN		BIT(1)
 #define DIG_CLK_CTL_RXD3_CLK_EN		BIT(2)
+#define DIG_CLK_CTL_D_MBHC_CLK_EN_MASK	BIT(3)
+#define DIG_CLK_CTL_D_MBHC_CLK_EN	BIT(3)
 #define DIG_CLK_CTL_TXD_CLK_EN		BIT(4)
 #define DIG_CLK_CTL_NCP_CLK_EN_MASK	BIT(6)
 #define DIG_CLK_CTL_NCP_CLK_EN		BIT(6)
@@ -132,8 +141,51 @@
 #define MICB_1_INT_TX3_INT_PULLUP_EN_TX1N_TO_GND	0
 
 #define CDC_A_MICB_2_EN			(0xf144)
+#define CDC_A_MICB_2_EN_ENABLE		BIT(7)
+#define CDC_A_MICB_2_PULL_DOWN_EN_MASK	BIT(5)
+#define CDC_A_MICB_2_PULL_DOWN_EN	BIT(5)
 #define CDC_A_TX_1_2_ATEST_CTL_2	(0xf145)
 #define CDC_A_MASTER_BIAS_CTL		(0xf146)
+#define CDC_A_MBHC_DET_CTL_1		(0xf147)
+#define CDC_A_MBHC_DET_CTL_L_DET_EN			BIT(7)
+#define CDC_A_MBHC_DET_CTL_GND_DET_EN			BIT(6)
+#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION	BIT(5)
+#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_REMOVAL	(0)
+#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_MASK		BIT(5)
+#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_SHIFT		(5)
+#define CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO		BIT(4)
+#define CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_MANUAL		BIT(3)
+#define CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_MASK		GENMASK(4, 3)
+#define CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN			BIT(2)
+#define CDC_A_MBHC_DET_CTL_2		(0xf150)
+#define CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0	(BIT(7) | BIT(6))
+#define CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD	BIT(5)
+#define CDC_A_PLUG_TYPE_MASK				GENMASK(4, 3)
+#define CDC_A_HPHL_PLUG_TYPE_NO				BIT(4)
+#define CDC_A_GND_PLUG_TYPE_NO				BIT(3)
+#define CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN_MASK	BIT(0)
+#define CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN		BIT(0)
+#define CDC_A_MBHC_FSM_CTL		(0xf151)
+#define CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN			BIT(7)
+#define CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN_MASK		BIT(7)
+#define CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_I_100UA	(0x3 << 4)
+#define CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_MASK		GENMASK(6, 4)
+#define CDC_A_MBHC_DBNC_TIMER		(0xf152)
+#define CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS		BIT(3)
+#define CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS	(0x9 << 4)
+#define CDC_A_MBHC_BTN0_ZDET_CTL_0	(0xf153)
+#define CDC_A_MBHC_BTN1_ZDET_CTL_1	(0xf154)
+#define CDC_A_MBHC_BTN2_ZDET_CTL_2	(0xf155)
+#define CDC_A_MBHC_BTN3_CTL		(0xf156)
+#define CDC_A_MBHC_BTN4_CTL		(0xf157)
+#define CDC_A_MBHC_BTN_VREF_FINE_SHIFT	(2)
+#define CDC_A_MBHC_BTN_VREF_FINE_MASK	GENMASK(4, 2)
+#define CDC_A_MBHC_BTN_VREF_COARSE_MASK	GENMASK(7, 5)
+#define CDC_A_MBHC_BTN_VREF_COARSE_SHIFT (5)
+#define CDC_A_MBHC_BTN_VREF_MASK	(CDC_A_MBHC_BTN_VREF_COARSE_MASK | \
+					CDC_A_MBHC_BTN_VREF_FINE_MASK)
+#define CDC_A_MBHC_RESULT_1		(0xf158)
+#define CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK	GENMASK(4, 0)
 #define CDC_A_TX_1_EN			(0xf160)
 #define CDC_A_TX_2_EN			(0xf161)
 #define CDC_A_TX_1_2_TEST_CTL_1		(0xf162)
@@ -217,16 +269,34 @@
 #define MSM8916_WCD_ANALOG_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
 				    SNDRV_PCM_FMTBIT_S24_LE)
 
+static int btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+	       SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4;
+static int hs_jack_mask = SND_JACK_HEADPHONE | SND_JACK_HEADSET;
+
 static const char * const supply_names[] = {
 	"vdd-cdc-io",
 	"vdd-cdc-tx-rx-cx",
 };
 
+#define MBHC_MAX_BUTTONS	(5)
+
 struct pm8916_wcd_analog_priv {
 	u16 pmic_rev;
 	u16 codec_version;
+	bool	mbhc_btn_enabled;
+	/* special event to detect accessory type */
+	bool	mbhc_btn0_pressed;
+	bool	detect_accessory_type;
 	struct clk *mclk;
+	struct snd_soc_codec *codec;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+	struct snd_soc_jack *jack;
+	bool hphl_jack_type_normally_open;
+	bool gnd_jack_type_normally_open;
+	/* Voltage threshold when internal current source of 100uA is used */
+	u32 vref_btn_cs[MBHC_MAX_BUTTONS];
+	/* Voltage threshold when microphone bias is ON */
+	u32 vref_btn_micb[MBHC_MAX_BUTTONS];
 	unsigned int micbias1_cap_mode;
 	unsigned int micbias2_cap_mode;
 	unsigned int micbias_mv;
@@ -373,6 +443,97 @@ static int pm8916_wcd_analog_enable_micbias_int1(struct
 						     wcd->micbias1_cap_mode);
 }
 
+static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd)
+{
+	struct snd_soc_codec *codec = wcd->codec;
+	u32 plug_type = 0;
+	u32 int_en_mask;
+
+	snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1,
+		      CDC_A_MBHC_DET_CTL_L_DET_EN |
+		      CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION |
+		      CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO |
+		      CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN);
+
+	if (wcd->hphl_jack_type_normally_open)
+		plug_type |= CDC_A_HPHL_PLUG_TYPE_NO;
+
+	if (wcd->gnd_jack_type_normally_open)
+		plug_type |= CDC_A_GND_PLUG_TYPE_NO;
+
+	snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2,
+		      CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 |
+		      CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD |
+		      plug_type |
+		      CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN);
+
+
+	snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER,
+		      CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS |
+		      CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS);
+
+	/* enable MBHC clock */
+	snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL,
+			    DIG_CLK_CTL_D_MBHC_CLK_EN_MASK,
+			    DIG_CLK_CTL_D_MBHC_CLK_EN);
+
+	int_en_mask = MBHC_SWITCH_INT;
+	if (wcd->mbhc_btn_enabled)
+		int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET;
+
+	snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0);
+	snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask);
+	wcd->mbhc_btn0_pressed = false;
+	wcd->detect_accessory_type = true;
+}
+
+static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv,
+				      bool micbias2_enabled)
+{
+	struct snd_soc_codec *codec = priv->codec;
+	u32 coarse, fine, reg_val, reg_addr;
+	int *vrefs, i;
+
+	if (!micbias2_enabled) { /* use internal 100uA Current source */
+		/* Enable internal 2.2k Internal Rbias Resistor */
+		snd_soc_update_bits(codec, CDC_A_MICB_1_INT_RBIAS,
+				    MICB_1_INT_TX2_INT_RBIAS_EN_MASK,
+				    MICB_1_INT_TX2_INT_RBIAS_EN_ENABLE);
+		/* Remove pull down on MIC BIAS2 */
+		snd_soc_update_bits(codec, CDC_A_MICB_2_EN,
+				   CDC_A_MICB_2_PULL_DOWN_EN_MASK,
+				   0);
+		/* enable 100uA internal current source */
+		snd_soc_update_bits(codec, CDC_A_MBHC_FSM_CTL,
+				    CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_MASK,
+				    CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_I_100UA);
+	}
+	snd_soc_update_bits(codec, CDC_A_MBHC_FSM_CTL,
+			CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN_MASK,
+			CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN);
+
+	if (micbias2_enabled)
+		vrefs = &priv->vref_btn_micb[0];
+	else
+		vrefs = &priv->vref_btn_cs[0];
+
+	/* program vref ranges for all the buttons */
+	reg_addr = CDC_A_MBHC_BTN0_ZDET_CTL_0;
+	for (i = 0; i <  MBHC_MAX_BUTTONS; i++) {
+		/* split mv in to coarse parts of 100mv & fine parts of 12mv */
+		coarse = (vrefs[i] / 100);
+		fine = ((vrefs[i] % 100) / 12);
+		reg_val = (coarse << CDC_A_MBHC_BTN_VREF_COARSE_SHIFT) |
+			 (fine << CDC_A_MBHC_BTN_VREF_FINE_SHIFT);
+		snd_soc_update_bits(codec, reg_addr,
+			       CDC_A_MBHC_BTN_VREF_MASK,
+			       reg_val);
+		reg_addr++;
+	}
+
+	return 0;
+}
+
 static int pm8916_wcd_analog_enable_micbias_int2(struct
 						  snd_soc_dapm_widget
 						  *w, struct snd_kcontrol
@@ -381,6 +542,15 @@ static int pm8916_wcd_analog_enable_micbias_int2(struct
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
 	struct pm8916_wcd_analog_priv *wcd = snd_soc_codec_get_drvdata(codec);
 
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		pm8916_mbhc_configure_bias(wcd, true);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		pm8916_mbhc_configure_bias(wcd, false);
+		break;
+	}
+
 	return pm8916_wcd_analog_enable_micbias_int(codec, event, w->reg,
 						     wcd->micbias2_cap_mode);
 }
@@ -548,9 +718,14 @@ static int pm8916_wcd_analog_probe(struct snd_soc_codec *codec)
 		snd_soc_write(codec, wcd_reg_defaults_2_0[reg].reg,
 			      wcd_reg_defaults_2_0[reg].def);
 
+	priv->codec = codec;
+
 	snd_soc_update_bits(codec, CDC_D_CDC_RST_CTL,
 			    RST_CTL_DIG_SW_RST_N_MASK,
 			    RST_CTL_DIG_SW_RST_N_REMOVE_RESET);
+
+	pm8916_wcd_setup_mbhc(priv);
+
 	return 0;
 }
 
@@ -749,11 +924,130 @@ static const struct snd_soc_dapm_widget pm8916_wcd_analog_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("A_MCLK2", CDC_D_CDC_TOP_CLK_CTL, 3, 0, NULL, 0),
 };
 
+static int pm8916_wcd_analog_set_jack(struct snd_soc_codec *codec,
+				      struct snd_soc_jack *jack,
+				      void *data)
+{
+	struct pm8916_wcd_analog_priv *wcd = snd_soc_codec_get_drvdata(codec);
+
+	wcd->jack = jack;
+
+	return 0;
+}
+
 static struct regmap *pm8916_get_regmap(struct device *dev)
 {
 	return dev_get_regmap(dev->parent, NULL);
 }
 
+static irqreturn_t mbhc_btn_release_irq_handler(int irq, void *arg)
+{
+	struct pm8916_wcd_analog_priv *priv = arg;
+
+	if (priv->detect_accessory_type) {
+		struct snd_soc_codec *codec = priv->codec;
+		u32 val = snd_soc_read(codec, CDC_A_MBHC_RESULT_1);
+
+		/* check if its BTN0 thats released */
+		if ((val >= 0) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK))
+			priv->mbhc_btn0_pressed = false;
+
+	} else {
+		snd_soc_jack_report(priv->jack, 0, btn_mask);
+	}
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t mbhc_btn_press_irq_handler(int irq, void *arg)
+{
+	struct pm8916_wcd_analog_priv *priv = arg;
+	struct snd_soc_codec *codec = priv->codec;
+	u32 btn_result;
+
+	btn_result = snd_soc_read(codec, CDC_A_MBHC_RESULT_1) &
+				  CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK;
+
+	switch (btn_result) {
+	case 0xf:
+		snd_soc_jack_report(priv->jack, SND_JACK_BTN_4, btn_mask);
+		break;
+	case 0x7:
+		snd_soc_jack_report(priv->jack, SND_JACK_BTN_3, btn_mask);
+		break;
+	case 0x3:
+		snd_soc_jack_report(priv->jack, SND_JACK_BTN_2, btn_mask);
+		break;
+	case 0x1:
+		snd_soc_jack_report(priv->jack, SND_JACK_BTN_1, btn_mask);
+		break;
+	case 0x0:
+		/* handle BTN_0 specially for type detection */
+		if (priv->detect_accessory_type)
+			priv->mbhc_btn0_pressed = true;
+		else
+			snd_soc_jack_report(priv->jack,
+					    SND_JACK_BTN_0, btn_mask);
+		break;
+	default:
+		dev_err(codec->dev,
+			"Unexpected button press result (%x)", btn_result);
+		break;
+	}
+
+	return IRQ_HANDLED;
+}
+
+
+static irqreturn_t pm8916_mbhc_switch_irq_handler(int irq, void *arg)
+{
+	struct pm8916_wcd_analog_priv *priv = arg;
+	struct snd_soc_codec *codec = priv->codec;
+	bool ins = false;
+
+	if (snd_soc_read(codec, CDC_A_MBHC_DET_CTL_1) &
+				CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_MASK)
+		ins = true;
+
+	/* Set the detection type appropriately */
+	snd_soc_update_bits(codec, CDC_A_MBHC_DET_CTL_1,
+			    CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_MASK,
+			    (!ins << CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_SHIFT));
+
+
+	if (ins) { /* hs insertion */
+		bool micbias_enabled = false;
+
+		if (snd_soc_read(codec, CDC_A_MICB_2_EN) &
+				CDC_A_MICB_2_EN_ENABLE)
+			micbias_enabled = true;
+
+		pm8916_mbhc_configure_bias(priv, micbias_enabled);
+
+		/*
+		 * if only a btn0 press event is receive just before
+		 * insert event then its a 3 pole headphone else if
+		 * both press and release event received then its
+		 * a headset.
+		 */
+		if (priv->mbhc_btn0_pressed)
+			snd_soc_jack_report(priv->jack,
+					    SND_JACK_HEADPHONE, hs_jack_mask);
+		else
+			snd_soc_jack_report(priv->jack,
+					    SND_JACK_HEADSET, hs_jack_mask);
+
+		priv->detect_accessory_type = false;
+
+	} else { /* removal */
+		snd_soc_jack_report(priv->jack, 0, hs_jack_mask);
+		priv->detect_accessory_type = true;
+		priv->mbhc_btn0_pressed = false;
+	}
+
+	return IRQ_HANDLED;
+}
+
 static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = {
 	[0] = {
 	       .name = "pm8916_wcd_analog_pdm_rx",
@@ -782,6 +1076,7 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = {
 static const struct snd_soc_codec_driver pm8916_wcd_analog = {
 	.probe = pm8916_wcd_analog_probe,
 	.remove = pm8916_wcd_analog_remove,
+	.set_jack = pm8916_wcd_analog_set_jack,
 	.get_regmap = pm8916_get_regmap,
 	.component_driver = {
 		.controls = pm8916_wcd_analog_snd_controls,
@@ -796,6 +1091,7 @@ static const struct snd_soc_codec_driver pm8916_wcd_analog = {
 static int pm8916_wcd_analog_parse_dt(struct device *dev,
 				       struct pm8916_wcd_analog_priv *priv)
 {
+	int rval;
 
 	if (of_property_read_bool(dev->of_node, "qcom,micbias1-ext-cap"))
 		priv->micbias1_cap_mode = MICB_1_EN_EXT_BYP_CAP;
@@ -810,6 +1106,39 @@ static int pm8916_wcd_analog_parse_dt(struct device *dev,
 	of_property_read_u32(dev->of_node, "qcom,micbias-lvl",
 			     &priv->micbias_mv);
 
+	if (of_property_read_bool(dev->of_node,
+				  "qcom,hphl-jack-type-normally-open"))
+		priv->hphl_jack_type_normally_open = true;
+	else
+		priv->hphl_jack_type_normally_open = false;
+
+	if (of_property_read_bool(dev->of_node,
+				  "qcom,gnd-jack-type-normally-open"))
+		priv->gnd_jack_type_normally_open = true;
+	else
+		priv->gnd_jack_type_normally_open = false;
+
+	priv->mbhc_btn_enabled = true;
+	rval = of_property_read_u32_array(dev->of_node,
+					  "qcom,mbhc-vthreshold-low",
+					  &priv->vref_btn_cs[0],
+					  MBHC_MAX_BUTTONS);
+	if (rval < 0) {
+		priv->mbhc_btn_enabled = false;
+	} else {
+		rval = of_property_read_u32_array(dev->of_node,
+						  "qcom,mbhc-vthreshold-high",
+						  &priv->vref_btn_micb[0],
+						  MBHC_MAX_BUTTONS);
+		if (rval < 0)
+			priv->mbhc_btn_enabled = false;
+	}
+
+	if (!priv->mbhc_btn_enabled)
+		dev_err(dev,
+			"DT property missing, MBHC btn detection disabled\n");
+
+
 	return 0;
 }
 
@@ -817,7 +1146,7 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 {
 	struct pm8916_wcd_analog_priv *priv;
 	struct device *dev = &pdev->dev;
-	int ret, i;
+	int ret, i, irq;
 
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
 	if (!priv)
@@ -849,6 +1178,48 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	irq = platform_get_irq_byname(pdev, "mbhc_switch_int");
+	if (irq < 0) {
+		dev_err(dev, "failed to get mbhc switch irq\n");
+		return irq;
+	}
+
+	ret = devm_request_irq(dev, irq, pm8916_mbhc_switch_irq_handler,
+			       IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING |
+			       IRQF_ONESHOT,
+			       "mbhc switch irq", priv);
+	if (ret)
+		dev_err(dev, "cannot request mbhc switch irq\n");
+
+	if (priv->mbhc_btn_enabled) {
+		irq = platform_get_irq_byname(pdev, "mbhc_but_press_det");
+		if (irq < 0) {
+			dev_err(dev, "failed to get button press irq\n");
+			return irq;
+		}
+
+		ret = devm_request_irq(dev, irq, mbhc_btn_press_irq_handler,
+				       IRQF_TRIGGER_RISING |
+				       IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
+				       "mbhc btn press irq", priv);
+		if (ret)
+			dev_err(dev, "cannot request mbhc button press irq\n");
+
+		irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det");
+		if (irq < 0) {
+			dev_err(dev, "failed to get button release irq\n");
+			return irq;
+		}
+
+		ret = devm_request_irq(dev, irq, mbhc_btn_release_irq_handler,
+				       IRQF_TRIGGER_RISING |
+				       IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
+				       "mbhc btn release irq", priv);
+		if (ret)
+			dev_err(dev, "cannot request mbhc button release irq\n");
+
+	}
+
 	dev_set_drvdata(dev, priv);
 
 	return snd_soc_register_codec(dev, &pm8916_wcd_analog,
-- 
cgit v1.2.3


From b47b91c8504c62f95ddff2876620d68697927bd4 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla
Date: Thu, 17 Aug 2017 10:02:11 +0200
Subject: ASoC: qcom: apq8016-sbc: Add support to Headset JACK

This patch adds support to Headset JACK, also provides board specific
vref ranges for mbhc buttons to be detected.

This headset supports both 3 pole and 4 pole headset type and 5 buttons.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/apq8016_sbc.c | 34 ++++++++++++++++++++++++++++++++++
 1 file changed, 34 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index 96a079d9f697..d49adc822a11 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -21,12 +21,16 @@
 #include <linux/platform_device.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
+#include <sound/jack.h>
 #include <sound/soc.h>
+#include <uapi/linux/input-event-codes.h>
 #include <dt-bindings/sound/apq8016-lpass.h>
 
 struct apq8016_sbc_data {
 	void __iomem *mic_iomux;
 	void __iomem *spkr_iomux;
+	struct snd_soc_jack jack;
+	bool jack_setup;
 	struct snd_soc_dai_link dai_link[];	/* dynamically allocated */
 };
 
@@ -70,6 +74,31 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
 
 	}
 
+	if (!pdata->jack_setup) {
+		struct snd_jack *jack;
+
+		rval = snd_soc_card_jack_new(card, "Headset Jack",
+					     SND_JACK_HEADSET |
+					     SND_JACK_HEADPHONE |
+					     SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+					     SND_JACK_BTN_2 | SND_JACK_BTN_3 |
+					     SND_JACK_BTN_4,
+					     &pdata->jack, NULL, 0);
+
+		if (rval < 0) {
+			dev_err(card->dev, "Unable to add Headphone Jack\n");
+			return rval;
+		}
+
+		jack = pdata->jack.jack;
+
+		snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_MEDIA);
+		snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+		snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+		snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+		pdata->jack_setup = true;
+	}
+
 	for (i = 0 ; i < dai_link->num_codecs; i++) {
 		struct snd_soc_dai *dai = rtd->codec_dais[i];
 
@@ -81,6 +110,11 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
 			dev_warn(card->dev, "Failed to set mclk: %d\n", rval);
 			return rval;
 		}
+		rval = snd_soc_codec_set_jack(codec, &pdata->jack, NULL);
+		if (rval != 0 && rval != -ENOTSUPP) {
+			dev_warn(card->dev, "Failed to set jack: %d\n", rval);
+			return rval;
+		}
 	}
 
 	return 0;
-- 
cgit v1.2.3


From 80b649039240ec41a6ab0a5e347853bc78abfd2d Mon Sep 17 00:00:00 2001
From: Harsha Priya N
Date: Mon, 21 Aug 2017 12:26:52 -0700
Subject: ASoC: Intel: Headset button support in kabylake machine driver

This patch adds headset button support for kabylake machine driver
(kbl_rt5663_rt5514_max98927).

Signed-off-by: Hsin-yu Chao <hychao@google.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index b51dff6fe25b..69681b04d7fc 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -18,6 +18,7 @@
  * GNU General Public License for more details.
  */
 
+#include <linux/input.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
@@ -176,6 +177,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
 	int ret;
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_jack *jack;
 
 	/*
 	 * Headset buttons map to the google Reference headset.
@@ -190,6 +192,12 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
 		return ret;
 	}
 
+	jack = &ctx->kabylake_headset;
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+
 	rt5663_set_jack_detect(codec, &ctx->kabylake_headset);
 
 	ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "DMIC");
-- 
cgit v1.2.3


From bd29ae969042c87413157c02d4f2a787d0212da3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:44 +0200
Subject: ASoC: jack: Manage gpios via devres

Let's be lazy -- this patch adds the devres code to
snd_soc_jack_add_gpios() for releasing the gpio resources at device
removal automagically.  After this patch, you don't have to call
snd_soc_jack_free_gpios() manually as long as it's managed from the
machine driver.

What about the gpios assigned in other levels?  Well, you might still
need to free the resources manually, depending on how the component
unbind works.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-jack.c | 52 +++++++++++++++++++++++++++++++++++++++++-----------
 1 file changed, 41 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 42ca9f19843b..4c83734ae596 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -22,6 +22,12 @@
 #include <linux/suspend.h>
 #include <trace/events/asoc.h>
 
+struct jack_gpio_tbl {
+	int count;
+	struct snd_soc_jack *jack;
+	struct snd_soc_jack_gpio *gpios;
+};
+
 /**
  * snd_soc_codec_set_jack - configure codec jack.
  * @codec: CODEC
@@ -333,6 +339,28 @@ static int snd_soc_jack_pm_notifier(struct notifier_block *nb,
 	return NOTIFY_DONE;
 }
 
+static void jack_free_gpios(struct snd_soc_jack *jack, int count,
+			    struct snd_soc_jack_gpio *gpios)
+{
+	int i;
+
+	for (i = 0; i < count; i++) {
+		gpiod_unexport(gpios[i].desc);
+		unregister_pm_notifier(&gpios[i].pm_notifier);
+		free_irq(gpiod_to_irq(gpios[i].desc), &gpios[i]);
+		cancel_delayed_work_sync(&gpios[i].work);
+		gpiod_put(gpios[i].desc);
+		gpios[i].jack = NULL;
+	}
+}
+
+static void jack_devres_free_gpios(struct device *dev, void *res)
+{
+	struct jack_gpio_tbl *tbl = res;
+
+	jack_free_gpios(tbl->jack, tbl->count, tbl->gpios);
+}
+
 /**
  * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack
  *
@@ -347,6 +375,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 			struct snd_soc_jack_gpio *gpios)
 {
 	int i, ret;
+	struct jack_gpio_tbl *tbl;
+
+	tbl = devres_alloc(jack_devres_free_gpios, sizeof(*tbl), GFP_KERNEL);
+	if (!tbl)
+		return -ENOMEM;
+	tbl->jack = jack;
+	tbl->count = count;
+	tbl->gpios = gpios;
 
 	for (i = 0; i < count; i++) {
 		if (!gpios[i].name) {
@@ -424,12 +460,14 @@ got_gpio:
 				      msecs_to_jiffies(gpios[i].debounce_time));
 	}
 
+	devres_add(jack->card->dev, tbl);
 	return 0;
 
 err:
 	gpio_free(gpios[i].gpio);
 undo:
-	snd_soc_jack_free_gpios(jack, i, gpios);
+	jack_free_gpios(jack, i, gpios);
+	devres_free(tbl);
 
 	return ret;
 }
@@ -471,16 +509,8 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpiods);
 void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
 			struct snd_soc_jack_gpio *gpios)
 {
-	int i;
-
-	for (i = 0; i < count; i++) {
-		gpiod_unexport(gpios[i].desc);
-		unregister_pm_notifier(&gpios[i].pm_notifier);
-		free_irq(gpiod_to_irq(gpios[i].desc), &gpios[i]);
-		cancel_delayed_work_sync(&gpios[i].work);
-		gpiod_put(gpios[i].desc);
-		gpios[i].jack = NULL;
-	}
+	jack_free_gpios(jack, count, gpios);
+	devres_destroy(jack->card->dev, jack_devres_free_gpios, NULL, NULL);
 }
 EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios);
 #endif	/* CONFIG_GPIOLIB */
-- 
cgit v1.2.3


From 8244fd7e489a586e63a36302105a2a1ece28a3c6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:45 +0200
Subject: ASoC: fsl: Remove superfluous snd_soc_jack_free_gpios() call

Since jack gpios are managed via devres, we don't have to call
snd_jack_free_gpios() at release any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/imx-es8328.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index 20e7400e2611..9953438086e4 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -203,9 +203,6 @@ static int imx_es8328_remove(struct platform_device *pdev)
 {
 	struct imx_es8328_data *data = platform_get_drvdata(pdev);
 
-	snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios),
-				headset_jack_gpios);
-
 	snd_soc_unregister_card(&data->card);
 
 	return 0;
-- 
cgit v1.2.3


From 2661112fcc5fe555130f383a21e6ce5152570403 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:46 +0200
Subject: ASoC: simple-card: Remove superfluous snd_soc_jack_free_gpios() call

Since jack gpios are managed via devres, we don't have to call
snd_jack_free_gpios() at release any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card.c | 10 ----------
 1 file changed, 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index dfaf48ff88b0..6959a74a6f49 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -104,12 +104,6 @@ static int asoc_simple_card_init_jack(struct snd_soc_card *card,
 	return 0;
 }
 
-static void asoc_simple_card_remove_jack(struct asoc_simple_jack *sjack)
-{
-	if (gpio_is_valid(sjack->gpio.gpio))
-		snd_soc_jack_free_gpios(&sjack->jack, 1, &sjack->gpio);
-}
-
 static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -493,10 +487,6 @@ err:
 static int asoc_simple_card_remove(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	struct simple_card_data *priv = snd_soc_card_get_drvdata(card);
-
-	asoc_simple_card_remove_jack(&priv->hp_jack);
-	asoc_simple_card_remove_jack(&priv->mic_jack);
 
 	return asoc_simple_card_clean_reference(card);
 }
-- 
cgit v1.2.3


From 51f25e7f9270378cee0526184ecaaae6c847a5f6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:47 +0200
Subject: ASoC: intel: Remove superfluous snd_soc_jack_free_gpios() call

Since jack gpios are managed via devres, we don't have to call
snd_jack_free_gpios() at release any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/byt-max98090.c | 12 ------------
 1 file changed, 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c
index 047be7fa0ce9..0f8b8209c020 100644
--- a/sound/soc/intel/boards/byt-max98090.c
+++ b/sound/soc/intel/boards/byt-max98090.c
@@ -173,20 +173,8 @@ static int byt_max98090_probe(struct platform_device *pdev)
 	return 0;
 }
 
-static int byt_max98090_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	struct byt_max98090_private *priv = snd_soc_card_get_drvdata(card);
-
-	snd_soc_jack_free_gpios(&priv->jack, ARRAY_SIZE(hs_jack_gpios),
-				hs_jack_gpios);
-
-	return 0;
-}
-
 static struct platform_driver byt_max98090_driver = {
 	.probe = byt_max98090_probe,
-	.remove = byt_max98090_remove,
 	.driver = {
 		.name = "byt-max98090",
 		.pm = &snd_soc_pm_ops,
-- 
cgit v1.2.3


From 90c265c65a3216ea236ed8e81d0a8d6411c26e88 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:48 +0200
Subject: ASoC: omap: Remove superfluous snd_soc_jack_free_gpios() call

Since jack gpios are managed via devres, we don't have to call
snd_jack_free_gpios() at release any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/omap/ams-delta.c    | 10 ----------
 sound/soc/omap/omap-twl4030.c | 13 -------------
 sound/soc/omap/rx51.c         |  9 ---------
 3 files changed, 32 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 16cc95fa4573..6c49f3d6fd96 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -513,15 +513,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int ams_delta_card_remove(struct snd_soc_card *card)
-{
-	snd_soc_jack_free_gpios(&ams_delta_hook_switch,
-			ARRAY_SIZE(ams_delta_hook_switch_gpios),
-			ams_delta_hook_switch_gpios);
-
-	return 0;
-}
-
 /* DAI glue - connects codec <--> CPU */
 static struct snd_soc_dai_link ams_delta_dai_link = {
 	.name = "CX20442",
@@ -540,7 +531,6 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
 static struct snd_soc_card ams_delta_audio_card = {
 	.name = "AMS_DELTA",
 	.owner = THIS_MODULE,
-	.remove = ams_delta_card_remove,
 	.dai_link = &ams_delta_dai_link,
 	.num_links = 1,
 
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index a24b0dedabb9..cccc316743fa 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -208,18 +208,6 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
-static int omap_twl4030_card_remove(struct snd_soc_card *card)
-{
-	struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card);
-
-	if (priv->jack_detect > 0)
-		snd_soc_jack_free_gpios(&priv->hs_jack,
-					ARRAY_SIZE(hs_jack_gpios),
-					hs_jack_gpios);
-
-	return 0;
-}
-
 /* Digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
 	{
@@ -247,7 +235,6 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
 /* Audio machine driver */
 static struct snd_soc_card omap_twl4030_card = {
 	.owner = THIS_MODULE,
-	.remove = omap_twl4030_card_remove,
 	.dai_link = omap_twl4030_dai_links,
 	.num_links = ARRAY_SIZE(omap_twl4030_dai_links),
 
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 3aeb65feaea1..57448bd5ad77 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -311,14 +311,6 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 	return err;
 }
 
-static int rx51_card_remove(struct snd_soc_card *card)
-{
-	snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
-				rx51_av_jack_gpios);
-
-	return 0;
-}
-
 /* Digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link rx51_dai[] = {
 	{
@@ -361,7 +353,6 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = {
 static struct snd_soc_card rx51_sound_card = {
 	.name = "RX-51",
 	.owner = THIS_MODULE,
-	.remove = rx51_card_remove,
 	.dai_link = rx51_dai,
 	.num_links = ARRAY_SIZE(rx51_dai),
 	.aux_dev = rx51_aux_dev,
-- 
cgit v1.2.3


From 7676b40606666547544998e5bc9e7f492fa2e13f Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:49 +0200
Subject: ASoC: pxa: Remove superfluous snd_soc_jack_free_gpios() call

Since jack gpios are managed via devres, we don't have to call
snd_jack_free_gpios() at release any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/pxa/hx4700.c | 8 --------
 1 file changed, 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index a9ac881c2e14..6cdef5d4954e 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -138,13 +138,6 @@ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
 	return err;
 }
 
-static int hx4700_card_remove(struct snd_soc_card *card)
-{
-	snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio);
-
-	return 0;
-}
-
 /* hx4700 digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link hx4700_dai = {
 	.name = "ak4641",
@@ -163,7 +156,6 @@ static struct snd_soc_dai_link hx4700_dai = {
 static struct snd_soc_card snd_soc_card_hx4700 = {
 	.name			= "iPAQ hx4700",
 	.owner			= THIS_MODULE,
-	.remove			= hx4700_card_remove,
 	.dai_link		= &hx4700_dai,
 	.num_links		= 1,
 	.dapm_widgets		= hx4700_dapm_widgets,
-- 
cgit v1.2.3


From 790b82898db2f42e3430ca84ac9bc3bd7ab474ce Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:50 +0200
Subject: ASoC: samsung: Remove superfluous snd_soc_jack_free_gpios() call

Since jack gpios are managed via devres, we don't have to call
snd_jack_free_gpios() at release any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/h1940_uda1380.c  |  9 ---------
 sound/soc/samsung/rx1950_uda1380.c | 10 ----------
 sound/soc/samsung/smartq_wm8987.c  |  9 ---------
 3 files changed, 28 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 5f5825faeb2a..051935162d7b 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -170,14 +170,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int h1940_uda1380_card_remove(struct snd_soc_card *card)
-{
-	snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
-		hp_jack_gpios);
-
-	return 0;
-}
-
 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link h1940_uda1380_dai[] = {
 	{
@@ -197,7 +189,6 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = {
 static struct snd_soc_card h1940_asoc = {
 	.name = "h1940",
 	.owner = THIS_MODULE,
-	.remove = h1940_uda1380_card_remove,
 	.dai_link = h1940_uda1380_dai,
 	.num_links = ARRAY_SIZE(h1940_uda1380_dai),
 
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index fa096abe9e75..a064ca7d78c3 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -31,7 +31,6 @@
 #include "s3c24xx-i2s.h"
 
 static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd);
-static int rx1950_uda1380_card_remove(struct snd_soc_card *card);
 static int rx1950_startup(struct snd_pcm_substream *substream);
 static int rx1950_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params);
@@ -118,7 +117,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 static struct snd_soc_card rx1950_asoc = {
 	.name = "rx1950",
 	.owner = THIS_MODULE,
-	.remove = rx1950_uda1380_card_remove,
 	.dai_link = rx1950_uda1380_dai,
 	.num_links = ARRAY_SIZE(rx1950_uda1380_dai),
 
@@ -219,14 +217,6 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int rx1950_uda1380_card_remove(struct snd_soc_card *card)
-{
-	snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
-		hp_jack_gpios);
-
-	return 0;
-}
-
 static int __init rx1950_init(void)
 {
 	int ret;
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index 425ee2ba37f0..cf0f54e652c1 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -160,14 +160,6 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd)
 	return err;
 }
 
-static int smartq_wm8987_card_remove(struct snd_soc_card *card)
-{
-	snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios),
-				smartq_jack_gpios);
-
-	return 0;
-}
-
 static struct snd_soc_dai_link smartq_dai[] = {
 	{
 		.name		= "wm8987",
@@ -186,7 +178,6 @@ static struct snd_soc_dai_link smartq_dai[] = {
 static struct snd_soc_card snd_soc_smartq = {
 	.name = "SmartQ",
 	.owner = THIS_MODULE,
-	.remove = smartq_wm8987_card_remove,
 	.dai_link = smartq_dai,
 	.num_links = ARRAY_SIZE(smartq_dai),
 
-- 
cgit v1.2.3


From 2a8bd83af30e3045b8b251191e759a2be9877816 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 22 Aug 2017 13:59:51 +0200
Subject: ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call

Since jack gpios are managed via devres, we don't have to call
snd_jack_free_gpios() at release any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra_alc5632.c  | 13 -------------
 sound/soc/tegra/tegra_max98090.c | 18 ------------------
 sound/soc/tegra/tegra_rt5640.c   | 13 -------------
 sound/soc/tegra/tegra_rt5677.c   | 18 ------------------
 sound/soc/tegra/tegra_wm8903.c   |  5 -----
 5 files changed, 67 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 0509902512cc..00383155d5f8 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -124,18 +124,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int tegra_alc5632_card_remove(struct snd_soc_card *card)
-{
-	struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card);
-
-	if (gpio_is_valid(machine->gpio_hp_det)) {
-		snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1,
-					&tegra_alc5632_hp_jack_gpio);
-	}
-
-	return 0;
-}
-
 static struct snd_soc_dai_link tegra_alc5632_dai = {
 	.name = "ALC5632",
 	.stream_name = "ALC5632 PCM",
@@ -150,7 +138,6 @@ static struct snd_soc_dai_link tegra_alc5632_dai = {
 static struct snd_soc_card snd_soc_tegra_alc5632 = {
 	.name = "tegra-alc5632",
 	.owner = THIS_MODULE,
-	.remove = tegra_alc5632_card_remove,
 	.dai_link = &tegra_alc5632_dai,
 	.num_links = 1,
 	.controls = tegra_alc5632_controls,
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
index c34a54d6e812..955067adf2d9 100644
--- a/sound/soc/tegra/tegra_max98090.c
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -176,23 +176,6 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int tegra_max98090_card_remove(struct snd_soc_card *card)
-{
-	struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card);
-
-	if (gpio_is_valid(machine->gpio_hp_det)) {
-		snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1,
-					&tegra_max98090_hp_jack_gpio);
-	}
-
-	if (gpio_is_valid(machine->gpio_mic_det)) {
-		snd_soc_jack_free_gpios(&tegra_max98090_mic_jack, 1,
-					&tegra_max98090_mic_jack_gpio);
-	}
-
-	return 0;
-}
-
 static struct snd_soc_dai_link tegra_max98090_dai = {
 	.name = "max98090",
 	.stream_name = "max98090 PCM",
@@ -206,7 +189,6 @@ static struct snd_soc_dai_link tegra_max98090_dai = {
 static struct snd_soc_card snd_soc_tegra_max98090 = {
 	.name = "tegra-max98090",
 	.owner = THIS_MODULE,
-	.remove = tegra_max98090_card_remove,
 	.dai_link = &tegra_max98090_dai,
 	.num_links = 1,
 	.controls = tegra_max98090_controls,
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index 93a356802345..76ed6042d84d 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -126,18 +126,6 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int tegra_rt5640_card_remove(struct snd_soc_card *card)
-{
-	struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card);
-
-	if (gpio_is_valid(machine->gpio_hp_det)) {
-		snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1,
-					&tegra_rt5640_hp_jack_gpio);
-	}
-
-	return 0;
-}
-
 static struct snd_soc_dai_link tegra_rt5640_dai = {
 	.name = "RT5640",
 	.stream_name = "RT5640 PCM",
@@ -151,7 +139,6 @@ static struct snd_soc_dai_link tegra_rt5640_dai = {
 static struct snd_soc_card snd_soc_tegra_rt5640 = {
 	.name = "tegra-rt5640",
 	.owner = THIS_MODULE,
-	.remove = tegra_rt5640_card_remove,
 	.dai_link = &tegra_rt5640_dai,
 	.num_links = 1,
 	.controls = tegra_rt5640_controls,
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index ebf58d0e0f10..dc5a295d61e5 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -169,23 +169,6 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int tegra_rt5677_card_remove(struct snd_soc_card *card)
-{
-	struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card);
-
-	if (gpio_is_valid(machine->gpio_hp_det)) {
-		snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1,
-				&tegra_rt5677_hp_jack_gpio);
-	}
-
-	if (gpio_is_valid(machine->gpio_mic_present)) {
-		snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1,
-				&tegra_rt5677_mic_jack_gpio);
-	}
-
-	return 0;
-}
-
 static struct snd_soc_dai_link tegra_rt5677_dai = {
 	.name = "RT5677",
 	.stream_name = "RT5677 PCM",
@@ -199,7 +182,6 @@ static struct snd_soc_dai_link tegra_rt5677_dai = {
 static struct snd_soc_card snd_soc_tegra_rt5677 = {
 	.name = "tegra-rt5677",
 	.owner = THIS_MODULE,
-	.remove = tegra_rt5677_card_remove,
 	.dai_link = &tegra_rt5677_dai,
 	.num_links = 1,
 	.controls = tegra_rt5677_controls,
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index dbfb49298ae8..ce498cc0914b 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -205,11 +205,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card)
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 
-	if (gpio_is_valid(machine->gpio_hp_det)) {
-		snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1,
-					&tegra_wm8903_hp_jack_gpio);
-	}
-
 	wm8903_mic_detect(codec, NULL, 0, 0);
 
 	return 0;
-- 
cgit v1.2.3


From 4958471b0d2110a34df0b45803e6f24ed89b857b Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Tue, 22 Aug 2017 15:57:21 +0800
Subject: ASoC: soc-core: Allow searching dai driver name in snd_soc_find_dai

Currently we are searching dai name in snd_soc_find_dai, which could
either be dai driver name or component device name(for legacy naming).

Allow searching dai driver name in snd_soc_find_dai too, so that we can
use dai driver name to find legacy naming dais.

Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 50c8dba54649..6fab0ff213ef 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1000,7 +1000,7 @@ static struct snd_soc_component *soc_find_component(
 /**
  * snd_soc_find_dai - Find a registered DAI
  *
- * @dlc: name of the DAI and optional component info to match
+ * @dlc: name of the DAI or the DAI driver and optional component info to match
  *
  * This function will search all registered components and their DAIs to
  * find the DAI of the same name. The component's of_node and name
@@ -1028,7 +1028,8 @@ struct snd_soc_dai *snd_soc_find_dai(
 		if (dlc->name && strcmp(component->name, dlc->name))
 			continue;
 		list_for_each_entry(dai, &component->dai_list, list) {
-			if (dlc->dai_name && strcmp(dai->name, dlc->dai_name))
+			if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)
+			    && strcmp(dai->driver->name, dlc->dai_name))
 				continue;
 
 			return dai;
-- 
cgit v1.2.3


From c16293560a1c188a05cba1dcc46548d33b641c06 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 22 Aug 2017 00:02:03 +0000
Subject: ASoC: rsnd: remove unused rsnd_xxx_to_dma()

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/src.c | 1 -
 sound/soc/sh/rcar/ssi.c | 1 -
 2 files changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 46400397dbbd..510b68a483b4 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -27,7 +27,6 @@ struct rsnd_src {
 #define RSND_SRC_NAME_SIZE 16
 
 #define rsnd_src_get(priv, id) ((struct rsnd_src *)(priv->src) + id)
-#define rsnd_src_to_dma(src) ((src)->dma)
 #define rsnd_src_nr(priv) ((priv)->src_nr)
 #define rsnd_src_sync_is_enabled(mod) (rsnd_mod_to_src(mod)->sen.val)
 
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 8bb47144660b..fffc07e72627 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -99,7 +99,6 @@ struct rsnd_ssi {
 	     i++)
 
 #define rsnd_ssi_get(priv, id) ((struct rsnd_ssi *)(priv->ssi) + id)
-#define rsnd_ssi_to_dma(mod) ((ssi)->dma)
 #define rsnd_ssi_nr(priv) ((priv)->ssi_nr)
 #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
 #define rsnd_ssi_flags_has(p, f) ((p)->flags & f)
-- 
cgit v1.2.3


From f628c4ed3de499f027d3ae70cbb46c6dfadfc098 Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Tue, 22 Aug 2017 23:35:48 +0800
Subject: ASoC: rockchip: Correct 'dmic-delay' property name

The 'dmic-delay' property name is different with the dt-binding.

So correct it with 'dmic-wakeup-delay-ms'.

Fixes: 3a6f9dce6116 (ASoC: rk3399_gru_sound: fix recording pop at first attempt)
Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rk3399_gru_sound.c | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index 70c0908fb9b4..566ccb39fb31 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -38,7 +38,7 @@
 
 #define SOUND_FS	256
 
-static unsigned int rt5514_dmic_delay;
+static unsigned int dmic_wakeup_delay;
 
 static struct snd_soc_jack rockchip_sound_jack;
 
@@ -126,7 +126,7 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* Wait for DMIC stable */
-	msleep(rt5514_dmic_delay);
+	msleep(dmic_wakeup_delay);
 
 	return 0;
 }
@@ -348,13 +348,13 @@ static int rockchip_sound_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	/* Set DMIC delay */
-	ret = device_property_read_u32(&pdev->dev, "dmic-delay",
-					&rt5514_dmic_delay);
+	/* Set DMIC wakeup delay */
+	ret = device_property_read_u32(&pdev->dev, "dmic-wakeup-delay-ms",
+					&dmic_wakeup_delay);
 	if (ret) {
-		rt5514_dmic_delay = 0;
+		dmic_wakeup_delay = 0;
 		dev_dbg(&pdev->dev,
-			"no optional property 'dmic-delay' found, default: no delay\n");
+			"no optional property 'dmic-wakeup-delay-ms' found, default: no delay\n");
 	}
 
 	rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL);
-- 
cgit v1.2.3


From f3f7df7635bd5fb6dbc8de150613d327a3bd6171 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Wed, 23 Aug 2017 08:03:43 +0200
Subject: ASoC: tegra: Fix unused variable warning

Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra_wm8903.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index ce498cc0914b..05f95944669a 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card)
 		snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 
 	wm8903_mic_detect(codec, NULL, 0, 0);
 
-- 
cgit v1.2.3


From 6cbcad079ecdb8ecd214d33688fd7f19167fe57a Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Wed, 23 Aug 2017 05:11:44 +0000
Subject: ASoC: wm8524: remove unnecessary snd_soc_unregister_platform()

wm8524 doesn't register platform, unregister platform is not needed

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8524.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c
index b66147e0631b..856a6950a451 100644
--- a/sound/soc/codecs/wm8524.c
+++ b/sound/soc/codecs/wm8524.c
@@ -233,10 +233,8 @@ static int wm8524_codec_probe(struct platform_device *pdev)
 
 	ret =  snd_soc_register_codec(&pdev->dev,
 			&soc_codec_dev_wm8524, &wm8524_dai, 1);
-	if (ret < 0) {
+	if (ret < 0)
 		dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
-		snd_soc_unregister_platform(&pdev->dev);
-	}
 
 	return ret;
 }
-- 
cgit v1.2.3


From b4e0180306f10a9d808b855dd77cdd0d7ae4f3fa Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 22 Aug 2017 04:56:06 +0000
Subject: ASoC: simple-scu-card: Add pm callbacks to platform driver

Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM
operations are going to happen. This is needed to get suspend/resume
working correctly when the audio is using simple-scu-card.

Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-scu-card.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 0bb3da755532..48606c63562a 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -300,6 +300,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match);
 static struct platform_driver asoc_simple_card = {
 	.driver = {
 		.name = "simple-scu-audio-card",
+		.pm = &snd_soc_pm_ops,
 		.of_match_table = asoc_simple_of_match,
 	},
 	.probe = asoc_simple_card_probe,
-- 
cgit v1.2.3


From 7b828a35ff834e45e5758a8a3f9d11db0526f9e1 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 22 Aug 2017 04:56:44 +0000
Subject: ASoC: audio-graph-card: Add pm callbacks to platform driver

Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM
operations are going to happen. This is needed to get suspend/resume
working correctly when the audio is using audio-graph-card.

Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 105ec3a6e30d..63048915efbb 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -325,6 +325,7 @@ MODULE_DEVICE_TABLE(of, asoc_graph_of_match);
 static struct platform_driver asoc_graph_card = {
 	.driver = {
 		.name = "asoc-audio-graph-card",
+		.pm = &snd_soc_pm_ops,
 		.of_match_table = asoc_graph_of_match,
 	},
 	.probe = asoc_graph_card_probe,
-- 
cgit v1.2.3


From 0995fb7c9802dd76422db384991f32296f8fbe1f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Tue, 22 Aug 2017 04:57:12 +0000
Subject: ASoC: audio-graph-scu-card: Add pm callbacks to platform driver

Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM
operations are going to happen. This is needed to get suspend/resume
working correctly when the audio is using audio-graph-scu-card.

Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-scu-card.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index dcd2df37bc3b..da73e5d06688 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -398,6 +398,7 @@ MODULE_DEVICE_TABLE(of, asoc_graph_of_match);
 static struct platform_driver asoc_graph_card = {
 	.driver = {
 		.name = "asoc-audio-graph-scu-card",
+		.pm = &snd_soc_pm_ops,
 		.of_match_table = asoc_graph_of_match,
 	},
 	.probe = asoc_graph_card_probe,
-- 
cgit v1.2.3


From 3b3011adada3bba47c56c205634e1b32512e0c7c Mon Sep 17 00:00:00 2001
From: Subhransu S. Prusty
Date: Tue, 22 Aug 2017 16:45:50 +0530
Subject: ASoC: Intel: Skylake: Fix to free dsp resource on ipc_init failure

For some dsp init error path, irq and few more resources are not freed.
This results in oops. So, fix it by freeing up the resources on ipc_init
failure.

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/bxt-sst.c | 4 +++-
 sound/soc/intel/skylake/cnl-sst.c | 4 +++-
 sound/soc/intel/skylake/skl-sst.c | 4 +++-
 3 files changed, 9 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index 5e1ac99ffa8b..4524211960e4 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -582,8 +582,10 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 			SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
 
 	ret = skl_ipc_init(dev, skl);
-	if (ret)
+	if (ret) {
+		skl_dsp_free(sst);
 		return ret;
+	}
 
 	/* set the D0i3 check */
 	skl->ipc.ops.check_dsp_lp_on = skl_ipc_check_D0i0;
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
index 48ef04d5da89..387de388ce29 100644
--- a/sound/soc/intel/skylake/cnl-sst.c
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -450,8 +450,10 @@ int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 			     CNL_ADSP_W1_SZ);
 
 	ret = cnl_ipc_init(dev, cnl);
-	if (ret)
+	if (ret) {
+		skl_dsp_free(sst);
 		return ret;
+	}
 
 	cnl->boot_complete = false;
 	init_waitqueue_head(&cnl->boot_wait);
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index a7ad734f591d..a436abf2fe3f 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -562,8 +562,10 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 			SKL_ADSP_W0_UP_SZ, SKL_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
 
 	ret = skl_ipc_init(dev, skl);
-	if (ret)
+	if (ret) {
+		skl_dsp_free(sst);
 		return ret;
+	}
 
 	sst->fw_ops = skl_fw_ops;
 
-- 
cgit v1.2.3


From f77d443c4c299aff5ad9c74811dd063f4d8bebcf Mon Sep 17 00:00:00 2001
From: Subhransu S. Prusty
Date: Tue, 22 Aug 2017 16:45:51 +0530
Subject: ASoC: Intel: Skylake: Fix to free resources for dsp_init failure

unmap mmio and free memory resources if dsp_init fails.

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 26 +++++++++++++++++++-------
 1 file changed, 19 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index bfb3332a77ca..f0f11f597b21 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -277,8 +277,10 @@ int skl_init_dsp(struct skl *skl)
 	}
 
 	ops = skl_get_dsp_ops(skl->pci->device);
-	if (!ops)
-		return -EIO;
+	if (!ops) {
+		goto unmap_mmio;
+		ret = -EIO;
+	}
 
 	loader_ops = ops->loader_ops();
 	ret = ops->init(bus->dev, mmio_base, irq,
@@ -286,25 +288,35 @@ int skl_init_dsp(struct skl *skl)
 				&skl->skl_sst);
 
 	if (ret < 0)
-		return ret;
+		goto unmap_mmio;
 
 	skl->skl_sst->dsp_ops = ops;
 	cores = &skl->skl_sst->cores;
 	cores->count = ops->num_cores;
 
 	cores->state = kcalloc(cores->count, sizeof(*cores->state), GFP_KERNEL);
-	if (!cores->state)
-		return -ENOMEM;
+	if (!cores->state) {
+		ret = -ENOMEM;
+		goto unmap_mmio;
+	}
 
 	cores->usage_count = kcalloc(cores->count, sizeof(*cores->usage_count),
 				     GFP_KERNEL);
 	if (!cores->usage_count) {
-		kfree(cores->state);
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto free_core_state;
 	}
 
 	dev_dbg(bus->dev, "dsp registration status=%d\n", ret);
 
+	return 0;
+
+free_core_state:
+	kfree(cores->state);
+
+unmap_mmio:
+	iounmap(mmio_base);
+
 	return ret;
 }
 
-- 
cgit v1.2.3


From c360e0c3ab8ef8d0fd6b5501a407b9a2be4f204d Mon Sep 17 00:00:00 2001
From: Subhransu S. Prusty
Date: Tue, 22 Aug 2017 16:45:52 +0530
Subject: ASoC: Intel: Skylake: Fix to free correct dev id in free_irq

The dev_id passed by the driver in request_threaded_irq is an ebus pointer,
whereas to free_irq it is hdac_bus. Fix by passing correct dev_id to
free_irq.

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index e612a4097275..d62ffbc93d54 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -415,7 +415,7 @@ static int skl_free(struct hdac_ext_bus *ebus)
 	snd_hdac_ext_stop_streams(ebus);
 
 	if (bus->irq >= 0)
-		free_irq(bus->irq, (void *)bus);
+		free_irq(bus->irq, (void *)ebus);
 	snd_hdac_bus_free_stream_pages(bus);
 	snd_hdac_stream_free_all(ebus);
 	snd_hdac_link_free_all(ebus);
-- 
cgit v1.2.3


From 7b992c24de78a206b3abd07192686d2d5db5012c Mon Sep 17 00:00:00 2001
From: Subhransu S. Prusty
Date: Tue, 22 Aug 2017 16:45:53 +0530
Subject: ASoC: Intel: Skylake: Fix DSP core ref count for init failure

During dsp init failure, the ref count is not incremented and dsp is
powered down. But as the skl driver calls put_core for the init failure it
decrements the dsp core ref count and ref count becomes unbalanced.

This results in dsp core powered up in further runtime suspend/resume
cycles and never powered down.

So increment the ref count before dsp core powerup and for any failure,
decrement in put_core will be balanced.

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-sst-dsp.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index c2ba5ff1632c..19ee1d4f3bdf 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -351,6 +351,8 @@ int skl_dsp_get_core(struct sst_dsp *ctx, unsigned int core_id)
 		return -EINVAL;
 	}
 
+	skl->cores.usage_count[core_id]++;
+
 	if (skl->cores.state[core_id] == SKL_DSP_RESET) {
 		ret = ctx->fw_ops.set_state_D0(ctx, core_id);
 		if (ret < 0) {
@@ -359,8 +361,6 @@ int skl_dsp_get_core(struct sst_dsp *ctx, unsigned int core_id)
 		}
 	}
 
-	skl->cores.usage_count[core_id]++;
-
 out:
 	dev_dbg(ctx->dev, "core id %d state %d usage_count %d\n",
 			core_id, skl->cores.state[core_id],
-- 
cgit v1.2.3


From cb0236eca79c8ceb5cdb8d2a15ce272e754d2921 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart
Date: Tue, 22 Aug 2017 15:32:35 -0500
Subject: ASoC: codecs: rt5645: add quirks for Asus T100HA

Fix jack detection and dmic setup

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 20 ++++++++++++++++++++
 1 file changed, 20 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 9ec58166f7c4..500bb42d0804 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3637,6 +3637,24 @@ static const struct dmi_system_id dmi_platform_gpd_win[] = {
 	{}
 };
 
+static struct rt5645_platform_data general_platform_data2 = {
+	.dmic1_data_pin = RT5645_DMIC_DATA_IN2N,
+	.dmic2_data_pin = RT5645_DMIC2_DISABLE,
+	.jd_mode = 3,
+	.inv_jd1_1 = true,
+};
+
+static struct dmi_system_id dmi_platform_asus_t100ha[] = {
+	{
+		.ident = "ASUS T100HAN",
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "T100HAN"),
+		},
+	},
+	{ }
+};
+
 static bool rt5645_check_dp(struct device *dev)
 {
 	if (device_property_present(dev, "realtek,in2-differential") ||
@@ -3689,6 +3707,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 		rt5645->pdata = general_platform_data;
 	else if (dmi_check_system(dmi_platform_gpd_win))
 		rt5645->pdata = gpd_win_platform_data;
+	else if (dmi_check_system(dmi_platform_asus_t100ha))
+		rt5645->pdata = general_platform_data2;
 
 	if (quirk != -1) {
 		rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk);
-- 
cgit v1.2.3


From c26d8389be4dab2bb5a8ac16211a9966bf67091a Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart
Date: Tue, 22 Aug 2017 15:32:37 -0500
Subject: ASoC: codecs: rt5670: add jack detection quirk for Dell Venue 5585

Jack is not detected by default, using jdmode=3 works fine

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 16 ++++++++++++++++
 1 file changed, 16 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 0f4e5bc4476f..4ea51b487b8c 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2851,6 +2851,17 @@ static const struct dmi_system_id dmi_platform_intel_bytcht_jdmode2[] = {
 	{}
 };
 
+static const struct dmi_system_id dmi_platform_intel_bytcht_jdmode3[] = {
+	{
+		.ident = "Dell Venue 8 Pro 5855",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5855"),
+		},
+	},
+	{}
+};
+
 static int rt5670_i2c_probe(struct i2c_client *i2c,
 		    const struct i2c_device_id *id)
 {
@@ -2880,6 +2891,11 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 		rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P;
 		rt5670->pdata.dev_gpio = true;
 		rt5670->pdata.jd_mode = 2;
+	} else if (dmi_check_system(dmi_platform_intel_bytcht_jdmode3)) {
+		rt5670->pdata.dmic_en = true;
+		rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P;
+		rt5670->pdata.dev_gpio = true;
+		rt5670->pdata.jd_mode = 3;
 	}
 
 	rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap);
-- 
cgit v1.2.3


From cd6d6477f130c861c42e6eddc060e7c880e94bf3 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal
Date: Wed, 23 Aug 2017 20:05:22 +0530
Subject: ASoC: rt5645: make rt5645_platform_data const

Make these const as they are only used during a copy operation.
Done using Coccinelle.

@match disable optional_qualifier@
identifier s;
@@
static struct rt5645_platform_data s = {...};

@ref@
position p;
identifier match.s;
@@
s@p

@good1@
position ref.p;
identifier match.s,f,c;
expression e;
@@
(
e = s@p
|
e = s@p.f
|
c(...,s@p.f,...)
|
c(...,s@p,...)
)

@bad depends on  !good1@
position ref.p;
identifier match.s;
@@
s@p

@depends on forall !bad disable optional_qualifier@
identifier match.s;
@@
static
+ const
struct rt5645_platform_data s;

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 500bb42d0804..d2f038001908 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3559,7 +3559,7 @@ static const struct acpi_device_id rt5645_acpi_match[] = {
 MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match);
 #endif
 
-static struct rt5645_platform_data general_platform_data = {
+static const struct rt5645_platform_data general_platform_data = {
 	.dmic1_data_pin = RT5645_DMIC1_DISABLE,
 	.dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
 	.jd_mode = 3,
@@ -3593,7 +3593,7 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
 	{ }
 };
 
-static struct rt5645_platform_data buddy_platform_data = {
+static const struct rt5645_platform_data buddy_platform_data = {
 	.dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
 	.dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
 	.jd_mode = 3,
@@ -3610,7 +3610,7 @@ static struct dmi_system_id dmi_platform_intel_broadwell[] = {
 	{ }
 };
 
-static struct rt5645_platform_data gpd_win_platform_data = {
+static const struct rt5645_platform_data gpd_win_platform_data = {
 	.jd_mode = 3,
 	.inv_jd1_1 = true,
 };
-- 
cgit v1.2.3


From 9149916f1e8c912c585ddebf12a98e99bd81aafe Mon Sep 17 00:00:00 2001
From: Subhransu S. Prusty
Date: Wed, 23 Aug 2017 19:37:12 +0530
Subject: ASoC: Intel: Skylake: Fix uninitialized return

On failure to get dsp_ops, dsp_init returns error without assigning ret. ret
is assigned in code path which will never be executed. Fix it.

Fixes: f77d443c4c29 ("ASoC: Intel: Skylake: Fix to free resources for dsp_init failure"
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index f0f11f597b21..697d07bc1b5a 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -278,8 +278,8 @@ int skl_init_dsp(struct skl *skl)
 
 	ops = skl_get_dsp_ops(skl->pci->device);
 	if (!ops) {
-		goto unmap_mmio;
 		ret = -EIO;
+		goto unmap_mmio;
 	}
 
 	loader_ops = ops->loader_ops();
-- 
cgit v1.2.3


From 56026714bd45fccda710114982789c5145285560 Mon Sep 17 00:00:00 2001
From: Dan Carpenter
Date: Thu, 24 Aug 2017 13:07:41 +0300
Subject: ASoC: codecs: msm8916-wcd-analog: always true condition

Static checkers complain that unsigned int val is always >= 0.  The
context is that snd_soc_read() returns -1U on error.  This is harmless
because we're checking that CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK is not
set, and it will always be set for -1U.  I could have just removed the
check against -1 but I preserved it because I thought it helped with
readability.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-analog.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index f834a639b350..cd4d60ef4c77 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -949,7 +949,7 @@ static irqreturn_t mbhc_btn_release_irq_handler(int irq, void *arg)
 		u32 val = snd_soc_read(codec, CDC_A_MBHC_RESULT_1);
 
 		/* check if its BTN0 thats released */
-		if ((val >= 0) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK))
+		if ((val != -1) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK))
 			priv->mbhc_btn0_pressed = false;
 
 	} else {
-- 
cgit v1.2.3


From d18420b0a0b8f8f51f55fc654b2e0eb86d8b41d7 Mon Sep 17 00:00:00 2001
From: Hsin-Yu Chao
Date: Wed, 23 Aug 2017 18:29:31 +0800
Subject: ASoC: rt5514: expose Hotword Model control

This change exposes mixer control 'Hotword Model' for switching
hotword model in runtime. This new function requires updated
firmware to load hotword model at address 0x4ff80000.

Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 52 +++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/rt5514.h |  3 +++
 2 files changed, 55 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index a694ce9fed86..0e239e8c7bf8 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -302,6 +302,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 	struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component);
 	struct snd_soc_codec *codec = rt5514->codec;
 	const struct firmware *fw = NULL;
+	int ret = 0;
 
 	if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled)
 		return 0;
@@ -338,6 +339,27 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 				fw = NULL;
 			}
 
+			if (rt5514->model_buf && rt5514->model_len) {
+				ret = rt5514_spi_burst_write(0x4ff80000,
+					rt5514->model_buf,
+					((rt5514->model_len / 8) + 1) * 8);
+				if (ret) {
+					dev_err(codec->dev,
+						"Model load failed %d\n", ret);
+					return ret;
+				}
+			} else {
+				request_firmware(&fw, RT5514_FIRMWARE3,
+						 codec->dev);
+				if (fw) {
+					rt5514_spi_burst_write(0x4ff80000,
+						fw->data,
+						((fw->size/8)+1)*8);
+					release_firmware(fw);
+					fw = NULL;
+				}
+			}
+
 			/* DSP run */
 			regmap_write(rt5514->i2c_regmap, 0x18002f00,
 				0x00055148);
@@ -352,6 +374,34 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol,
+		const unsigned int __user *bytes, unsigned int size)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component);
+	struct snd_soc_codec *codec = rt5514->codec;
+	int ret = 0;
+
+	if (rt5514->model_buf || rt5514->model_len < size) {
+		if (rt5514->model_buf)
+			devm_kfree(codec->dev, rt5514->model_buf);
+		rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL);
+		if (!rt5514->model_buf) {
+			ret = -ENOMEM;
+			goto done;
+		}
+	}
+
+	/* Skips the TLV header. */
+	bytes += 2;
+
+	if (copy_from_user(rt5514->model_buf, bytes, size))
+		ret = -EFAULT;
+done:
+	rt5514->model_len = (ret ? 0 : size);
+	return ret;
+}
+
 static const struct snd_kcontrol_new rt5514_snd_controls[] = {
 	SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST,
 		RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv),
@@ -363,6 +413,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = {
 		adc_vol_tlv),
 	SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0,
 		rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put),
+	SND_SOC_BYTES_TLV("Hotword Model", 0x8504,
+		NULL, rt5514_hotword_model_put),
 };
 
 /* ADC Mixer*/
diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h
index 2dc40e6d8b3f..803311cb7e2a 100644
--- a/sound/soc/codecs/rt5514.h
+++ b/sound/soc/codecs/rt5514.h
@@ -255,6 +255,7 @@
 
 #define RT5514_FIRMWARE1	"rt5514_dsp_fw1.bin"
 #define RT5514_FIRMWARE2	"rt5514_dsp_fw2.bin"
+#define RT5514_FIRMWARE3	"rt5514_dsp_fw3.bin"
 
 /* System Clock Source */
 enum {
@@ -281,6 +282,8 @@ struct rt5514_priv {
 	int pll_in;
 	int pll_out;
 	int dsp_enabled;
+	u8 *model_buf;
+	unsigned int model_len;
 };
 
 #endif /* __RT5514_H__ */
-- 
cgit v1.2.3


From 6a6dafda937cfcdbbfe46a3e093de8bb929ba52d Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Thu, 24 Aug 2017 12:40:17 +0800
Subject: ASoC: Add a sanity check before using dai driver name

The dai driver's name is allowed to be NULL. So add a sanity check for
that.

Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Reported-by: Donglin Peng <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6fab0ff213ef..74c17068bb11 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1029,7 +1029,8 @@ struct snd_soc_dai *snd_soc_find_dai(
 			continue;
 		list_for_each_entry(dai, &component->dai_list, list) {
 			if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)
-			    && strcmp(dai->driver->name, dlc->dai_name))
+			    && (!dai->driver->name
+				|| strcmp(dai->driver->name, dlc->dai_name)))
 				continue;
 
 			return dai;
-- 
cgit v1.2.3


From 9fc129f6e5de01ce5fccf033485e40dfa429b45c Mon Sep 17 00:00:00 2001
From: Shreyas NC
Date: Wed, 23 Aug 2017 19:33:48 +0530
Subject: ASoC: Intel: Skylake: Parse multiple manifest data blocks

Currently we can parse a single manifest data block. But manifest
private data can have multiple data blocks.

So, fix the parsing logic to parse multiple data blocks by returning
offset of each parsed data block.

Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 12 +++++-------
 1 file changed, 5 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 350d2387f737..f14b373f65d7 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -2721,14 +2721,12 @@ static int skl_tplg_get_manifest_tkn(struct device *dev,
 
 			tkn_count = tkn_count + ret;
 			tkn_elem++;
-			tuple_size += tkn_count *
-				sizeof(struct snd_soc_tplg_vendor_value_elem);
-			break;
 		}
+		tuple_size += (tkn_count * sizeof(*tkn_elem));
 		tkn_count = 0;
 	}
 
-	return 0;
+	return off;
 }
 
 /*
@@ -2751,11 +2749,10 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest,
 	num_blocks = ret;
 
 	off += array->size;
-	array = (struct snd_soc_tplg_vendor_array *)
-			(manifest->priv.data + off);
-
 	/* Read the BLOCK_TYPE and BLOCK_SIZE descriptor */
 	while (num_blocks > 0) {
+		array = (struct snd_soc_tplg_vendor_array *)
+				(manifest->priv.data + off);
 		ret = skl_tplg_get_desc_blocks(dev, array);
 
 		if (ret < 0)
@@ -2789,6 +2786,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest,
 		} else {
 			return -EINVAL;
 		}
+		off += ret;
 	}
 
 	return 0;
-- 
cgit v1.2.3


From ca312fda690e94c704bcb0e21b73b5d5bd1cf1ff Mon Sep 17 00:00:00 2001
From: Shreyas NC
Date: Wed, 23 Aug 2017 19:33:50 +0530
Subject: ASoC: Intel: Skylake: Commonize parsing of format tokens

Format resource tokens can be a part of either the widget or manifest
private data. In the current model, format resources come as a part of
widget private data and they come as a part of topology manifest in the
newly introduced model.

So add a common function that can fill up either of the structures.

Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 47 +++++++++++++++++++---------------
 1 file changed, 26 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index f14b373f65d7..df808fbf14b5 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -1948,27 +1948,9 @@ static int skl_tplg_fill_pins_info(struct device *dev,
  * on the direction
  */
 static int skl_tplg_fill_fmt(struct device *dev,
-		struct skl_module_cfg *mconfig,	u32 tkn,
-		u32 value, u32 dir, u32 pin_count)
+		struct skl_module_fmt *dst_fmt,
+		u32 tkn, u32 value)
 {
-	struct skl_module_fmt *dst_fmt;
-
-	switch (dir) {
-	case SKL_DIR_IN:
-		dst_fmt = mconfig->in_fmt;
-		dst_fmt += pin_count;
-		break;
-
-	case SKL_DIR_OUT:
-		dst_fmt = mconfig->out_fmt;
-		dst_fmt += pin_count;
-		break;
-
-	default:
-		dev_err(dev, "Invalid direction value\n");
-		return -EINVAL;
-	}
-
 	switch (tkn) {
 	case SKL_TKN_U32_FMT_CH:
 		dst_fmt->channels  = value;
@@ -2010,6 +1992,29 @@ static int skl_tplg_fill_fmt(struct device *dev,
 	return 0;
 }
 
+static int skl_tplg_widget_fill_fmt(struct device *dev,
+		struct skl_module_cfg *mconfig,
+		u32 tkn, u32 val, u32 dir, int fmt_idx)
+{
+	struct skl_module_fmt *dst_fmt;
+
+	switch (dir) {
+	case SKL_DIR_IN:
+		dst_fmt = &mconfig->in_fmt[fmt_idx];
+		break;
+
+	case SKL_DIR_OUT:
+		dst_fmt = &mconfig->out_fmt[fmt_idx];
+		break;
+
+	default:
+		dev_err(dev, "Invalid direction: %d\n", dir);
+		return -EINVAL;
+	}
+
+	return skl_tplg_fill_fmt(dev, dst_fmt, tkn, val);
+}
+
 static int skl_tplg_get_uuid(struct device *dev, struct skl_module_cfg *mconfig,
 	      struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn)
 {
@@ -2190,7 +2195,7 @@ static int skl_tplg_get_token(struct device *dev,
 	case SKL_TKN_U32_FMT_INTERLEAVE:
 	case SKL_TKN_U32_FMT_SAMPLE_TYPE:
 	case SKL_TKN_U32_FMT_CH_MAP:
-		ret = skl_tplg_fill_fmt(dev, mconfig, tkn_elem->token,
+		ret = skl_tplg_widget_fill_fmt(dev, mconfig, tkn_elem->token,
 				tkn_elem->value, dir, pin_index);
 
 		if (ret < 0)
-- 
cgit v1.2.3


From 822c3b044f7fbecc9894fcd3d19fc2ad51232fdb Mon Sep 17 00:00:00 2001
From: Shreyas NC
Date: Wed, 23 Aug 2017 19:33:51 +0530
Subject: ASoC: Intel: Skylake: Add driver structures to be filled from
 topology manifest

The topology manifest would include module common data including resource
and interface table. The resource table consists of resources required by
the dsp module such as buffer size, cycles per second, number of
input/output pins. And, the interface table consists of pcm parameters per
module which can be referenced later.

So define the structures accordingly to represent topology manifest data in
the driver.

Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.h | 52 ++++++++++++++++++++++++++++++++++
 sound/soc/intel/skylake/skl.h          |  2 ++
 2 files changed, 54 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index c25e8868b84e..93b2ba8c40f7 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -44,6 +44,13 @@
 #define SKL_DEFAULT_MIC_SEL_GAIN	0x3FF
 #define SKL_MIC_SEL_SWITCH	0x3
 
+#define SKL_OUTPUT_PIN		0
+#define SKL_INPUT_PIN		1
+#define SKL_MAX_PATH_CONFIGS	8
+#define SKL_MAX_MODULES_IN_PIPE	8
+#define SKL_MAX_MODULE_FORMATS		32
+#define SKL_MAX_MODULE_RESOURCES	32
+
 enum skl_channel_index {
 	SKL_CHANNEL_LEFT = 0,
 	SKL_CHANNEL_RIGHT = 1,
@@ -292,6 +299,51 @@ enum d0i3_capability {
 	SKL_D0I3_NON_STREAMING = 2,
 };
 
+struct skl_module_pin_fmt {
+	u8 id;
+	struct skl_module_fmt fmt;
+};
+
+struct skl_module_iface {
+	u8 fmt_idx;
+	u8 nr_in_fmt;
+	u8 nr_out_fmt;
+	struct skl_module_pin_fmt inputs[MAX_IN_QUEUE];
+	struct skl_module_pin_fmt outputs[MAX_OUT_QUEUE];
+};
+
+struct skl_module_pin_resources {
+	u8 pin_index;
+	u32 buf_size;
+};
+
+struct skl_module_res {
+	u8 id;
+	u32 is_pages;
+	u32 cps;
+	u32 ibs;
+	u32 obs;
+	u32 dma_buffer_size;
+	u32 cpc;
+	u8 nr_input_pins;
+	u8 nr_output_pins;
+	struct skl_module_pin_resources input[MAX_IN_QUEUE];
+	struct skl_module_pin_resources output[MAX_OUT_QUEUE];
+};
+
+struct skl_module {
+	uuid_le uuid;
+	u8 loadable;
+	u8 input_pin_type;
+	u8 output_pin_type;
+	u8 max_input_pins;
+	u8 max_output_pins;
+	u8 nr_resources;
+	u8 nr_interfaces;
+	struct skl_module_res resources[SKL_MAX_MODULE_RESOURCES];
+	struct skl_module_iface formats[SKL_MAX_MODULE_FORMATS];
+};
+
 struct skl_module_cfg {
 	u8 guid[16];
 	struct skl_module_inst_id id;
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 20017ee4685c..8d9d6899f761 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -71,6 +71,8 @@ struct skl {
 	struct work_struct probe_work;
 
 	struct skl_debug *debugfs;
+	u8 nr_modules;
+	struct skl_module **modules;
 };
 
 #define skl_to_ebus(s)	(&(s)->ebus)
-- 
cgit v1.2.3


From db6ed55de6da1125ff349bfe4ef12d18c366e893 Mon Sep 17 00:00:00 2001
From: Shreyas NC
Date: Wed, 23 Aug 2017 19:33:52 +0530
Subject: ASoC: Intel: Skylake: Populate module data from topology manifest

All the module common data will now be populated in the topology
manifest. This includes the resource and interface list supported by
the module. With this, driver need not compute the resources required
by each dsp module for a particular pcm parameter since it comes as a
part of the topology manifest.

So, add functions to parse the manifest tokens to populate the module
config data structure.

Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 320 ++++++++++++++++++++++++++++++++-
 1 file changed, 316 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index df808fbf14b5..f39883067f63 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -2037,6 +2037,104 @@ static void skl_tplg_fill_pin_dynamic_val(
 		mpin[i].is_dynamic = value;
 }
 
+/*
+ * Resource table in the manifest has pin specific resources
+ * like pin and pin buffer size
+ */
+static int skl_tplg_manifest_pin_res_tkn(struct device *dev,
+		struct snd_soc_tplg_vendor_value_elem *tkn_elem,
+		struct skl_module_res *res, int pin_idx, int dir)
+{
+	struct skl_module_pin_resources *m_pin;
+
+	switch (dir) {
+	case SKL_DIR_IN:
+		m_pin = &res->input[pin_idx];
+		break;
+
+	case SKL_DIR_OUT:
+		m_pin = &res->output[pin_idx];
+		break;
+
+	default:
+		dev_err(dev, "Invalid pin direction: %d\n", dir);
+		return -EINVAL;
+	}
+
+	switch (tkn_elem->token) {
+	case SKL_TKN_MM_U32_RES_PIN_ID:
+		m_pin->pin_index = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U32_PIN_BUF:
+		m_pin->buf_size = tkn_elem->value;
+		break;
+
+	default:
+		dev_err(dev, "Invalid token: %d\n", tkn_elem->token);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Fill module specific resources from the manifest's resource
+ * table like CPS, DMA size, mem_pages.
+ */
+static int skl_tplg_fill_res_tkn(struct device *dev,
+		struct snd_soc_tplg_vendor_value_elem *tkn_elem,
+		struct skl_module_res *res,
+		int pin_idx, int dir)
+{
+	int ret, tkn_count = 0;
+
+	if (!res)
+		return -EINVAL;
+
+	switch (tkn_elem->token) {
+	case SKL_TKN_MM_U32_CPS:
+		res->cps = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U32_DMA_SIZE:
+		res->dma_buffer_size = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U32_CPC:
+		res->cpc = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U32_MEM_PAGES:
+		res->is_pages = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U32_OBS:
+		res->obs = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U32_IBS:
+		res->ibs = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U32_RES_PIN_ID:
+	case SKL_TKN_MM_U32_PIN_BUF:
+		ret = skl_tplg_manifest_pin_res_tkn(dev, tkn_elem, res,
+						    pin_idx, dir);
+		if (ret < 0)
+			return ret;
+		break;
+
+	default:
+		dev_err(dev, "Not a res type token: %d", tkn_elem->token);
+		return -EINVAL;
+
+	}
+	tkn_count++;
+
+	return tkn_count;
+}
+
 /*
  * Parse tokens to fill up the module private data
  */
@@ -2625,13 +2723,13 @@ static int skl_tplg_fill_str_mfest_tkn(struct device *dev,
 			str_elem->string,
 			ARRAY_SIZE(skl->skl_sst->lib_info[ref_count].name));
 		ref_count++;
-		tkn_count++;
 		break;
 
 	default:
 		dev_err(dev, "Not a string token %d\n", str_elem->token);
 		break;
 	}
+	tkn_count++;
 
 	return tkn_count;
 }
@@ -2657,26 +2755,236 @@ static int skl_tplg_get_str_tkn(struct device *dev,
 	return tkn_count;
 }
 
+static int skl_tplg_manifest_fill_fmt(struct device *dev,
+		struct skl_module_iface *fmt,
+		struct snd_soc_tplg_vendor_value_elem *tkn_elem,
+		u32 dir, int fmt_idx)
+{
+	struct skl_module_pin_fmt *dst_fmt;
+	struct skl_module_fmt *mod_fmt;
+	int ret;
+
+	if (!fmt)
+		return -EINVAL;
+
+	switch (dir) {
+	case SKL_DIR_IN:
+		dst_fmt = &fmt->inputs[fmt_idx];
+		break;
+
+	case SKL_DIR_OUT:
+		dst_fmt = &fmt->outputs[fmt_idx];
+		break;
+
+	default:
+		dev_err(dev, "Invalid direction: %d\n", dir);
+		return -EINVAL;
+	}
+
+	mod_fmt = &dst_fmt->fmt;
+
+	switch (tkn_elem->token) {
+	case SKL_TKN_MM_U32_INTF_PIN_ID:
+		dst_fmt->id = tkn_elem->value;
+		break;
+
+	default:
+		ret = skl_tplg_fill_fmt(dev, mod_fmt, tkn_elem->token,
+					tkn_elem->value);
+		if (ret < 0)
+			return ret;
+		break;
+	}
+
+	return 0;
+}
+
+static int skl_tplg_fill_mod_info(struct device *dev,
+		struct snd_soc_tplg_vendor_value_elem *tkn_elem,
+		struct skl_module *mod)
+{
+
+	if (!mod)
+		return -EINVAL;
+
+	switch (tkn_elem->token) {
+	case SKL_TKN_U8_IN_PIN_TYPE:
+		mod->input_pin_type = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U8_OUT_PIN_TYPE:
+		mod->output_pin_type = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U8_IN_QUEUE_COUNT:
+		mod->max_input_pins = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U8_OUT_QUEUE_COUNT:
+		mod->max_output_pins = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U8_NUM_RES:
+		mod->nr_resources = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U8_NUM_INTF:
+		mod->nr_interfaces = tkn_elem->value;
+		break;
+
+	default:
+		dev_err(dev, "Invalid mod info token %d", tkn_elem->token);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+
 static int skl_tplg_get_int_tkn(struct device *dev,
 		struct snd_soc_tplg_vendor_value_elem *tkn_elem,
 		struct skl *skl)
 {
-	int tkn_count = 0;
+	int tkn_count = 0, ret;
+	static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx;
+	struct skl_module_res *res = NULL;
+	struct skl_module_iface *fmt = NULL;
+	struct skl_module *mod = NULL;
+	int i;
+
+	if (skl->modules) {
+		mod = skl->modules[mod_idx];
+		res = &mod->resources[res_val_idx];
+		fmt = &mod->formats[intf_val_idx];
+	}
 
 	switch (tkn_elem->token) {
 	case SKL_TKN_U32_LIB_COUNT:
 		skl->skl_sst->lib_count = tkn_elem->value;
-		tkn_count++;
+		break;
+
+	case SKL_TKN_U8_NUM_MOD:
+		skl->nr_modules = tkn_elem->value;
+		skl->modules = devm_kcalloc(dev, skl->nr_modules,
+				sizeof(*skl->modules), GFP_KERNEL);
+		if (!skl->modules)
+			return -ENOMEM;
+
+		for (i = 0; i < skl->nr_modules; i++) {
+			skl->modules[i] = devm_kzalloc(dev,
+					sizeof(struct skl_module), GFP_KERNEL);
+			if (!skl->modules[i])
+				return -ENOMEM;
+		}
+		break;
+
+	case SKL_TKN_MM_U8_MOD_IDX:
+		mod_idx = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U8_IN_PIN_TYPE:
+	case SKL_TKN_U8_OUT_PIN_TYPE:
+	case SKL_TKN_U8_IN_QUEUE_COUNT:
+	case SKL_TKN_U8_OUT_QUEUE_COUNT:
+	case SKL_TKN_MM_U8_NUM_RES:
+	case SKL_TKN_MM_U8_NUM_INTF:
+		ret = skl_tplg_fill_mod_info(dev, tkn_elem, mod);
+		if (ret < 0)
+			return ret;
+		break;
+
+	case SKL_TKN_U32_DIR_PIN_COUNT:
+		dir = tkn_elem->value & SKL_IN_DIR_BIT_MASK;
+		pin_idx = (tkn_elem->value & SKL_PIN_COUNT_MASK) >> 4;
+		break;
+
+	case SKL_TKN_MM_U32_RES_ID:
+		if (!res)
+			return -EINVAL;
+
+		res->id = tkn_elem->value;
+		res_val_idx = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U32_FMT_ID:
+		if (!fmt)
+			return -EINVAL;
+
+		fmt->fmt_idx = tkn_elem->value;
+		intf_val_idx = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U32_CPS:
+	case SKL_TKN_MM_U32_DMA_SIZE:
+	case SKL_TKN_MM_U32_CPC:
+	case SKL_TKN_U32_MEM_PAGES:
+	case SKL_TKN_U32_OBS:
+	case SKL_TKN_U32_IBS:
+	case SKL_TKN_MM_U32_RES_PIN_ID:
+	case SKL_TKN_MM_U32_PIN_BUF:
+		ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, pin_idx, dir);
+		if (ret < 0)
+			return ret;
+
+		break;
+
+	case SKL_TKN_MM_U32_NUM_IN_FMT:
+		if (!fmt)
+			return -EINVAL;
+
+		res->nr_input_pins = tkn_elem->value;
+		break;
+
+	case SKL_TKN_MM_U32_NUM_OUT_FMT:
+		if (!fmt)
+			return -EINVAL;
+
+		res->nr_output_pins = tkn_elem->value;
+		break;
+
+	case SKL_TKN_U32_FMT_CH:
+	case SKL_TKN_U32_FMT_FREQ:
+	case SKL_TKN_U32_FMT_BIT_DEPTH:
+	case SKL_TKN_U32_FMT_SAMPLE_SIZE:
+	case SKL_TKN_U32_FMT_CH_CONFIG:
+	case SKL_TKN_U32_FMT_INTERLEAVE:
+	case SKL_TKN_U32_FMT_SAMPLE_TYPE:
+	case SKL_TKN_U32_FMT_CH_MAP:
+	case SKL_TKN_MM_U32_INTF_PIN_ID:
+		ret = skl_tplg_manifest_fill_fmt(dev, fmt, tkn_elem,
+						 dir, pin_idx);
+		if (ret < 0)
+			return ret;
 		break;
 
 	default:
 		dev_err(dev, "Not a manifest token %d\n", tkn_elem->token);
 		return -EINVAL;
 	}
+	tkn_count++;
 
 	return tkn_count;
 }
 
+static int skl_tplg_get_manifest_uuid(struct device *dev,
+				struct skl *skl,
+				struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn)
+{
+	static int ref_count;
+	struct skl_module *mod;
+
+	if (uuid_tkn->token == SKL_TKN_UUID) {
+		mod = skl->modules[ref_count];
+		memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid));
+		ref_count++;
+	} else {
+		dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 /*
  * Fill the manifest structure by parsing the tokens based on the
  * type.
@@ -2709,7 +3017,11 @@ static int skl_tplg_get_manifest_tkn(struct device *dev,
 			continue;
 
 		case SND_SOC_TPLG_TUPLE_TYPE_UUID:
-			dev_warn(dev, "no uuid tokens for skl tplf manifest\n");
+			ret = skl_tplg_get_manifest_uuid(dev, skl, array->uuid);
+			if (ret < 0)
+				return ret;
+
+			tuple_size += sizeof(*array->uuid);
 			continue;
 
 		default:
-- 
cgit v1.2.3


From f6fa56e22559ade7287ca7c1b7218a08f516f379 Mon Sep 17 00:00:00 2001
From: Ramesh Babu
Date: Wed, 23 Aug 2017 19:33:53 +0530
Subject: ASoC: Intel: Skylake: Parse and update module config structure

A dsp path and the modules in the path can support various pcm
configurations. The list of supported pcm configurations from topology
manifest would be stored and later selected runtime based on the hw
pcm params. For legacy, module data is filled in the 0th index of
resource and interface table.

To accommodate both models, change the relevant structures and populate
them by parsing newly defined tokens. This change is backward compatible
with the existing model where driver computes the resources required by
each dsp module.

Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c |  59 ++++--
 sound/soc/intel/skylake/skl-pcm.c      |   3 +-
 sound/soc/intel/skylake/skl-topology.c | 342 +++++++++++++++++++++++++--------
 sound/soc/intel/skylake/skl-topology.h |  25 +++
 4 files changed, 334 insertions(+), 95 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 697d07bc1b5a..ea83e8bfb136 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -443,9 +443,12 @@ static void skl_set_base_module_format(struct skl_sst *ctx,
 			struct skl_module_cfg *mconfig,
 			struct skl_base_cfg *base_cfg)
 {
-	struct skl_module_fmt *format = &mconfig->in_fmt[0];
+	struct skl_module *module = mconfig->module;
+	struct skl_module_res *res = &module->resources[mconfig->res_idx];
+	struct skl_module_iface *fmt = &module->formats[mconfig->fmt_idx];
+	struct skl_module_fmt *format = &fmt->inputs[0].fmt;
 
-	base_cfg->audio_fmt.number_of_channels = (u8)format->channels;
+	base_cfg->audio_fmt.number_of_channels = format->channels;
 
 	base_cfg->audio_fmt.s_freq = format->s_freq;
 	base_cfg->audio_fmt.bit_depth = format->bit_depth;
@@ -460,10 +463,10 @@ static void skl_set_base_module_format(struct skl_sst *ctx,
 
 	base_cfg->audio_fmt.interleaving = format->interleaving_style;
 
-	base_cfg->cps = mconfig->mcps;
-	base_cfg->ibs = mconfig->ibs;
-	base_cfg->obs = mconfig->obs;
-	base_cfg->is_pages = mconfig->mem_pages;
+	base_cfg->cps = res->cps;
+	base_cfg->ibs = res->ibs;
+	base_cfg->obs = res->obs;
+	base_cfg->is_pages = res->is_pages;
 }
 
 /*
@@ -551,6 +554,9 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 			struct skl_cpr_cfg *cpr_mconfig)
 {
 	u32 dma_io_buf;
+	struct skl_module_res *res;
+	int res_idx = mconfig->res_idx;
+	struct skl *skl = get_skl_ctx(ctx->dev);
 
 	cpr_mconfig->gtw_cfg.node_id = skl_get_node_id(ctx, mconfig);
 
@@ -559,19 +565,27 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 		return;
 	}
 
+	if (skl->nr_modules) {
+		res = &mconfig->module->resources[mconfig->res_idx];
+		cpr_mconfig->gtw_cfg.dma_buffer_size = res->dma_buffer_size;
+		goto skip_buf_size_calc;
+	} else {
+		res = &mconfig->module->resources[res_idx];
+	}
+
 	switch (mconfig->hw_conn_type) {
 	case SKL_CONN_SOURCE:
 		if (mconfig->dev_type == SKL_DEVICE_HDAHOST)
-			dma_io_buf =  mconfig->ibs;
+			dma_io_buf =  res->ibs;
 		else
-			dma_io_buf =  mconfig->obs;
+			dma_io_buf =  res->obs;
 		break;
 
 	case SKL_CONN_SINK:
 		if (mconfig->dev_type == SKL_DEVICE_HDAHOST)
-			dma_io_buf =  mconfig->obs;
+			dma_io_buf =  res->obs;
 		else
-			dma_io_buf =  mconfig->ibs;
+			dma_io_buf =  res->ibs;
 		break;
 
 	default:
@@ -586,11 +600,12 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 	/* fallback to 2ms default value */
 	if (!cpr_mconfig->gtw_cfg.dma_buffer_size) {
 		if (mconfig->hw_conn_type == SKL_CONN_SOURCE)
-			cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs;
+			cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * res->obs;
 		else
-			cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs;
+			cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * res->ibs;
 	}
 
+skip_buf_size_calc:
 	cpr_mconfig->cpr_feature_mask = 0;
 	cpr_mconfig->gtw_cfg.config_length  = 0;
 
@@ -638,7 +653,9 @@ static void skl_setup_out_format(struct skl_sst *ctx,
 			struct skl_module_cfg *mconfig,
 			struct skl_audio_data_format *out_fmt)
 {
-	struct skl_module_fmt *format = &mconfig->out_fmt[0];
+	struct skl_module *module = mconfig->module;
+	struct skl_module_iface *fmt = &module->formats[mconfig->fmt_idx];
+	struct skl_module_fmt *format = &fmt->outputs[0].fmt;
 
 	out_fmt->number_of_channels = (u8)format->channels;
 	out_fmt->s_freq = format->s_freq;
@@ -663,7 +680,9 @@ static void skl_set_src_format(struct skl_sst *ctx,
 			struct skl_module_cfg *mconfig,
 			struct skl_src_module_cfg *src_mconfig)
 {
-	struct skl_module_fmt *fmt = &mconfig->out_fmt[0];
+	struct skl_module *module = mconfig->module;
+	struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx];
+	struct skl_module_fmt *fmt = &iface->outputs[0].fmt;
 
 	skl_set_base_module_format(ctx, mconfig,
 		(struct skl_base_cfg *)src_mconfig);
@@ -680,7 +699,9 @@ static void skl_set_updown_mixer_format(struct skl_sst *ctx,
 			struct skl_module_cfg *mconfig,
 			struct skl_up_down_mixer_cfg *mixer_mconfig)
 {
-	struct skl_module_fmt *fmt = &mconfig->out_fmt[0];
+	struct skl_module *module = mconfig->module;
+	struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx];
+	struct skl_module_fmt *fmt = &iface->outputs[0].fmt;
 	int i = 0;
 
 	skl_set_base_module_format(ctx,	mconfig,
@@ -1013,8 +1034,8 @@ int skl_unbind_modules(struct skl_sst *ctx,
 	struct skl_ipc_bind_unbind_msg msg;
 	struct skl_module_inst_id src_id = src_mcfg->id;
 	struct skl_module_inst_id dst_id = dst_mcfg->id;
-	int in_max = dst_mcfg->max_in_queue;
-	int out_max = src_mcfg->max_out_queue;
+	int in_max = dst_mcfg->module->max_input_pins;
+	int out_max = src_mcfg->module->max_output_pins;
 	int src_index, dst_index, src_pin_state, dst_pin_state;
 
 	skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
@@ -1075,8 +1096,8 @@ int skl_bind_modules(struct skl_sst *ctx,
 {
 	int ret;
 	struct skl_ipc_bind_unbind_msg msg;
-	int in_max = dst_mcfg->max_in_queue;
-	int out_max = src_mcfg->max_out_queue;
+	int in_max = dst_mcfg->module->max_input_pins;
+	int out_max = src_mcfg->module->max_output_pins;
 	int src_index, dst_index;
 
 	skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index f6c9adb6faa2..a756882b9029 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -1233,7 +1233,8 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig)
 	list_for_each_entry(module, &ctx->uuid_list, list) {
 		if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) {
 			mconfig->id.module_id = module->id;
-			mconfig->is_loadable = module->is_loadable;
+			if (mconfig->module)
+				mconfig->module->loadable = module->is_loadable;
 			return 0;
 		}
 	}
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index f39883067f63..22f768ca3c73 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -49,6 +49,9 @@ static const int mic_quatro_list[][SKL_CH_QUATRO] = {
 {0, 1, 2, 3},
 };
 
+#define CHECK_HW_PARAMS(ch, freq, bps, prm_ch, prm_freq, prm_bps) \
+	((ch == prm_ch) && (bps == prm_bps) && (freq == prm_freq))
+
 void skl_tplg_d0i3_get(struct skl *skl, enum d0i3_capability caps)
 {
 	struct skl_d0i3_data *d0i3 =  &skl->skl_sst->d0i3;
@@ -153,8 +156,10 @@ static bool skl_is_pipe_mcps_avail(struct skl *skl,
 				struct skl_module_cfg *mconfig)
 {
 	struct skl_sst *ctx = skl->skl_sst;
+	u8 res_idx = mconfig->res_idx;
+	struct skl_module_res *res = &mconfig->module->resources[res_idx];
 
-	if (skl->resource.mcps + mconfig->mcps > skl->resource.max_mcps) {
+	if (skl->resource.mcps + res->cps > skl->resource.max_mcps) {
 		dev_err(ctx->dev,
 			"%s: module_id %d instance %d\n", __func__,
 			mconfig->id.module_id, mconfig->id.instance_id);
@@ -170,7 +175,10 @@ static bool skl_is_pipe_mcps_avail(struct skl *skl,
 static void skl_tplg_alloc_pipe_mcps(struct skl *skl,
 				struct skl_module_cfg *mconfig)
 {
-	skl->resource.mcps += mconfig->mcps;
+	u8 res_idx = mconfig->res_idx;
+	struct skl_module_res *res = &mconfig->module->resources[res_idx];
+
+	skl->resource.mcps += res->cps;
 }
 
 /*
@@ -179,7 +187,11 @@ static void skl_tplg_alloc_pipe_mcps(struct skl *skl,
 static void
 skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig)
 {
-	skl->resource.mcps -= mconfig->mcps;
+	u8 res_idx = mconfig->res_idx;
+	struct skl_module_res *res = &mconfig->module->resources[res_idx];
+
+	res = &mconfig->module->resources[res_idx];
+	skl->resource.mcps -= res->cps;
 }
 
 /*
@@ -195,17 +207,21 @@ skl_tplg_free_pipe_mem(struct skl *skl, struct skl_module_cfg *mconfig)
 static void skl_dump_mconfig(struct skl_sst *ctx,
 					struct skl_module_cfg *mcfg)
 {
+	struct skl_module_iface *iface = &mcfg->module->formats[0];
+
 	dev_dbg(ctx->dev, "Dumping config\n");
 	dev_dbg(ctx->dev, "Input Format:\n");
-	dev_dbg(ctx->dev, "channels = %d\n", mcfg->in_fmt[0].channels);
-	dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->in_fmt[0].s_freq);
-	dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->in_fmt[0].ch_cfg);
-	dev_dbg(ctx->dev, "valid bit depth = %d\n", mcfg->in_fmt[0].valid_bit_depth);
+	dev_dbg(ctx->dev, "channels = %d\n", iface->inputs[0].fmt.channels);
+	dev_dbg(ctx->dev, "s_freq = %d\n", iface->inputs[0].fmt.s_freq);
+	dev_dbg(ctx->dev, "ch_cfg = %d\n", iface->inputs[0].fmt.ch_cfg);
+	dev_dbg(ctx->dev, "valid bit depth = %d\n",
+				iface->inputs[0].fmt.valid_bit_depth);
 	dev_dbg(ctx->dev, "Output Format:\n");
-	dev_dbg(ctx->dev, "channels = %d\n", mcfg->out_fmt[0].channels);
-	dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->out_fmt[0].s_freq);
-	dev_dbg(ctx->dev, "valid bit depth = %d\n", mcfg->out_fmt[0].valid_bit_depth);
-	dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg);
+	dev_dbg(ctx->dev, "channels = %d\n", iface->outputs[0].fmt.channels);
+	dev_dbg(ctx->dev, "s_freq = %d\n", iface->outputs[0].fmt.s_freq);
+	dev_dbg(ctx->dev, "valid bit depth = %d\n",
+				iface->outputs[0].fmt.valid_bit_depth);
+	dev_dbg(ctx->dev, "ch_cfg = %d\n", iface->outputs[0].fmt.ch_cfg);
 }
 
 static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs)
@@ -273,8 +289,8 @@ static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg,
 	struct skl_module_fmt *in_fmt, *out_fmt;
 
 	/* Fixups will be applied to pin 0 only */
-	in_fmt = &m_cfg->in_fmt[0];
-	out_fmt = &m_cfg->out_fmt[0];
+	in_fmt = &m_cfg->module->formats[0].inputs[0].fmt;
+	out_fmt = &m_cfg->module->formats[0].outputs[0].fmt;
 
 	if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (is_fe) {
@@ -312,21 +328,23 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx,
 {
 	int multiplier = 1;
 	struct skl_module_fmt *in_fmt, *out_fmt;
+	struct skl_module_res *res;
 
 	/* Since fixups is applied to pin 0 only, ibs, obs needs
 	 * change for pin 0 only
 	 */
-	in_fmt = &mcfg->in_fmt[0];
-	out_fmt = &mcfg->out_fmt[0];
+	res = &mcfg->module->resources[0];
+	in_fmt = &mcfg->module->formats[0].inputs[0].fmt;
+	out_fmt = &mcfg->module->formats[0].outputs[0].fmt;
 
 	if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT)
 		multiplier = 5;
 
-	mcfg->ibs = DIV_ROUND_UP(in_fmt->s_freq, 1000) *
+	res->ibs = DIV_ROUND_UP(in_fmt->s_freq, 1000) *
 			in_fmt->channels * (in_fmt->bit_depth >> 3) *
 			multiplier;
 
-	mcfg->obs = DIV_ROUND_UP(out_fmt->s_freq, 1000) *
+	res->obs = DIV_ROUND_UP(out_fmt->s_freq, 1000) *
 			out_fmt->channels * (out_fmt->bit_depth >> 3) *
 			multiplier;
 }
@@ -365,6 +383,8 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w,
 	struct nhlt_specific_cfg *cfg;
 	struct skl *skl = get_skl_ctx(ctx->dev);
 	u8 dev_type = skl_tplg_be_dev_type(m_cfg->dev_type);
+	int fmt_idx = m_cfg->fmt_idx;
+	struct skl_module_iface *m_iface = &m_cfg->module->formats[fmt_idx];
 
 	/* check if we already have blob */
 	if (m_cfg->formats_config.caps_size > 0)
@@ -375,23 +395,23 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w,
 	case SKL_DEVICE_DMIC:
 		link_type = NHLT_LINK_DMIC;
 		dir = SNDRV_PCM_STREAM_CAPTURE;
-		s_freq = m_cfg->in_fmt[0].s_freq;
-		s_fmt = m_cfg->in_fmt[0].bit_depth;
-		ch = m_cfg->in_fmt[0].channels;
+		s_freq = m_iface->inputs[0].fmt.s_freq;
+		s_fmt = m_iface->inputs[0].fmt.bit_depth;
+		ch = m_iface->inputs[0].fmt.channels;
 		break;
 
 	case SKL_DEVICE_I2S:
 		link_type = NHLT_LINK_SSP;
 		if (m_cfg->hw_conn_type == SKL_CONN_SOURCE) {
 			dir = SNDRV_PCM_STREAM_PLAYBACK;
-			s_freq = m_cfg->out_fmt[0].s_freq;
-			s_fmt = m_cfg->out_fmt[0].bit_depth;
-			ch = m_cfg->out_fmt[0].channels;
+			s_freq = m_iface->outputs[0].fmt.s_freq;
+			s_fmt = m_iface->outputs[0].fmt.bit_depth;
+			ch = m_iface->outputs[0].fmt.channels;
 		} else {
 			dir = SNDRV_PCM_STREAM_CAPTURE;
-			s_freq = m_cfg->in_fmt[0].s_freq;
-			s_fmt = m_cfg->in_fmt[0].bit_depth;
-			ch = m_cfg->in_fmt[0].channels;
+			s_freq = m_iface->inputs[0].fmt.s_freq;
+			s_fmt = m_iface->inputs[0].fmt.bit_depth;
+			ch = m_iface->inputs[0].fmt.channels;
 		}
 		break;
 
@@ -549,6 +569,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 	struct snd_soc_dapm_widget *w;
 	struct skl_module_cfg *mconfig;
 	struct skl_sst *ctx = skl->skl_sst;
+	u8 cfg_idx;
 	int ret = 0;
 
 	list_for_each_entry(w_module, &pipe->w_list, node) {
@@ -564,11 +585,15 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 			return -EIO;
 		}
 
+		cfg_idx = mconfig->pipe->cur_config_idx;
+		mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx;
+		mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx;
+
 		/* check resource available */
 		if (!skl_is_pipe_mcps_avail(skl, mconfig))
 			return -ENOMEM;
 
-		if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) {
+		if (mconfig->module->loadable && ctx->dsp->fw_ops.load_mod) {
 			ret = ctx->dsp->fw_ops.load_mod(ctx->dsp,
 				mconfig->id.module_id, mconfig->guid);
 			if (ret < 0)
@@ -633,7 +658,7 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
 		mconfig  = w_module->w->priv;
 		uuid_mod = (uuid_le *)mconfig->guid;
 
-		if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod &&
+		if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod &&
 			mconfig->m_state > SKL_MODULE_UNINIT) {
 			ret = ctx->dsp->fw_ops.unload_mod(ctx->dsp,
 						mconfig->id.module_id);
@@ -654,6 +679,65 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
 	return ret;
 }
 
+/*
+ * Here, we select pipe format based on the pipe type and pipe
+ * direction to determine the current config index for the pipeline.
+ * The config index is then used to select proper module resources.
+ * Intermediate pipes currently have a fixed format hence we select the
+ * 0th configuratation by default for such pipes.
+ */
+static int
+skl_tplg_get_pipe_config(struct skl *skl, struct skl_module_cfg *mconfig)
+{
+	struct skl_sst *ctx = skl->skl_sst;
+	struct skl_pipe *pipe = mconfig->pipe;
+	struct skl_pipe_params *params = pipe->p_params;
+	struct skl_path_config *pconfig = &pipe->configs[0];
+	struct skl_pipe_fmt *fmt = NULL;
+	bool in_fmt = false;
+	int i;
+
+	if (pipe->nr_cfgs == 0) {
+		pipe->cur_config_idx = 0;
+		return 0;
+	}
+
+	if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE) {
+		dev_dbg(ctx->dev, "No conn_type detected, take 0th config\n");
+		pipe->cur_config_idx = 0;
+		pipe->memory_pages = pconfig->mem_pages;
+
+		return 0;
+	}
+
+	if ((pipe->conn_type == SKL_PIPE_CONN_TYPE_FE &&
+	     pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) ||
+	     (pipe->conn_type == SKL_PIPE_CONN_TYPE_BE &&
+	     pipe->direction == SNDRV_PCM_STREAM_CAPTURE))
+		in_fmt = true;
+
+	for (i = 0; i < pipe->nr_cfgs; i++) {
+		pconfig = &pipe->configs[i];
+		if (in_fmt)
+			fmt = &pconfig->in_fmt;
+		else
+			fmt = &pconfig->out_fmt;
+
+		if (CHECK_HW_PARAMS(params->ch, params->s_freq, params->s_fmt,
+				    fmt->channels, fmt->freq, fmt->bps)) {
+			pipe->cur_config_idx = i;
+			pipe->memory_pages = pconfig->mem_pages;
+			dev_dbg(ctx->dev, "Using pipe config: %d\n", i);
+
+			return 0;
+		}
+	}
+
+	dev_err(ctx->dev, "Invalid pipe config: %d %d %d for pipe: %d\n",
+		params->ch, params->s_freq, params->s_fmt, pipe->ppl_id);
+	return -EINVAL;
+}
+
 /*
  * Mixer module represents a pipeline. So in the Pre-PMU event of mixer we
  * need create the pipeline. So we do following:
@@ -673,6 +757,10 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	struct skl_sst *ctx = skl->skl_sst;
 	struct skl_module_deferred_bind *modules;
 
+	ret = skl_tplg_get_pipe_config(skl, mconfig);
+	if (ret < 0)
+		return ret;
+
 	/* check resource available */
 	if (!skl_is_pipe_mcps_avail(skl, mconfig))
 		return -EBUSY;
@@ -776,12 +864,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
 	 * check all out/in pins are in bind state.
 	 * if so set the module param
 	 */
-	for (i = 0; i < mcfg->max_out_queue; i++) {
+	for (i = 0; i < mcfg->module->max_output_pins; i++) {
 		if (mcfg->m_out_pin[i].pin_state != SKL_PIN_BIND_DONE)
 			return 0;
 	}
 
-	for (i = 0; i < mcfg->max_in_queue; i++) {
+	for (i = 0; i < mcfg->module->max_input_pins; i++) {
 		if (mcfg->m_in_pin[i].pin_state != SKL_PIN_BIND_DONE)
 			return 0;
 	}
@@ -832,7 +920,7 @@ static int skl_tplg_module_add_deferred_bind(struct skl *skl,
 	int i;
 
 	/* only supported for module with static pin connection */
-	for (i = 0; i < dst->max_in_queue; i++) {
+	for (i = 0; i < dst->module->max_input_pins; i++) {
 		struct skl_module_pin *pin = &dst->m_in_pin[i];
 
 		if (pin->is_dynamic)
@@ -1092,7 +1180,7 @@ static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
 	if (ret)
 		return ret;
 
-	for (i = 0; i < sink_mconfig->max_in_queue; i++) {
+	for (i = 0; i < sink_mconfig->module->max_input_pins; i++) {
 		if (sink_mconfig->m_in_pin[i].pin_state == SKL_PIN_BIND_DONE) {
 			src_mconfig = sink_mconfig->m_in_pin[i].tgt_mcfg;
 			if (!src_mconfig)
@@ -1203,7 +1291,7 @@ static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 	if (ret)
 		return ret;
 
-	for (i = 0; i < src_mconfig->max_out_queue; i++) {
+	for (i = 0; i < src_mconfig->module->max_output_pins; i++) {
 		if (src_mconfig->m_out_pin[i].pin_state == SKL_PIN_BIND_DONE) {
 			sink_mconfig = src_mconfig->m_out_pin[i].tgt_mcfg;
 			if (!sink_mconfig)
@@ -1497,14 +1585,22 @@ int skl_tplg_update_pipe_params(struct device *dev,
 			struct skl_module_cfg *mconfig,
 			struct skl_pipe_params *params)
 {
+	struct skl_module_res *res = &mconfig->module->resources[0];
+	struct skl *skl = get_skl_ctx(dev);
 	struct skl_module_fmt *format = NULL;
+	u8 cfg_idx = mconfig->pipe->cur_config_idx;
 
 	skl_tplg_fill_dma_id(mconfig, params);
+	mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx;
+	mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx;
+
+	if (skl->nr_modules)
+		return 0;
 
 	if (params->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		format = &mconfig->in_fmt[0];
+		format = &mconfig->module->formats[0].inputs[0].fmt;
 	else
-		format = &mconfig->out_fmt[0];
+		format = &mconfig->module->formats[0].outputs[0].fmt;
 
 	/* set the hw_params */
 	format->s_freq = params->s_freq;
@@ -1532,11 +1628,11 @@ int skl_tplg_update_pipe_params(struct device *dev,
 	}
 
 	if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		mconfig->ibs = (format->s_freq / 1000) *
+		res->ibs = (format->s_freq / 1000) *
 				(format->channels) *
 				(format->bit_depth >> 3);
 	} else {
-		mconfig->obs = (format->s_freq / 1000) *
+		res->obs = (format->s_freq / 1000) *
 				(format->channels) *
 				(format->bit_depth >> 3);
 	}
@@ -1810,6 +1906,54 @@ static const struct snd_soc_tplg_kcontrol_ops skl_tplg_kcontrol_ops[] = {
 	},
 };
 
+static int skl_tplg_fill_pipe_cfg(struct device *dev,
+			struct skl_pipe *pipe, u32 tkn,
+			u32 tkn_val, int conf_idx, int dir)
+{
+	struct skl_pipe_fmt *fmt;
+	struct skl_path_config *config;
+
+	switch (dir) {
+	case SKL_DIR_IN:
+		fmt = &pipe->configs[conf_idx].in_fmt;
+		break;
+
+	case SKL_DIR_OUT:
+		fmt = &pipe->configs[conf_idx].out_fmt;
+		break;
+
+	default:
+		dev_err(dev, "Invalid direction: %d\n", dir);
+		return -EINVAL;
+	}
+
+	config = &pipe->configs[conf_idx];
+
+	switch (tkn) {
+	case SKL_TKN_U32_CFG_FREQ:
+		fmt->freq = tkn_val;
+		break;
+
+	case SKL_TKN_U8_CFG_CHAN:
+		fmt->channels = tkn_val;
+		break;
+
+	case SKL_TKN_U8_CFG_BPS:
+		fmt->bps = tkn_val;
+		break;
+
+	case SKL_TKN_U32_PATH_MEM_PGS:
+		config->mem_pages = tkn_val;
+		break;
+
+	default:
+		dev_err(dev, "Invalid token config: %d\n", tkn);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static int skl_tplg_fill_pipe_tkn(struct device *dev,
 			struct skl_pipe *pipe, u32 tkn,
 			u32 tkn_val)
@@ -1832,6 +1976,14 @@ static int skl_tplg_fill_pipe_tkn(struct device *dev,
 		pipe->lp_mode = tkn_val;
 		break;
 
+	case SKL_TKN_U32_PIPE_DIRECTION:
+		pipe->direction = tkn_val;
+		break;
+
+	case SKL_TKN_U32_NUM_CONFIGS:
+		pipe->nr_cfgs = tkn_val;
+		break;
+
 	default:
 		dev_err(dev, "Token not handled %d\n", tkn);
 		return -EINVAL;
@@ -1993,18 +2145,21 @@ static int skl_tplg_fill_fmt(struct device *dev,
 }
 
 static int skl_tplg_widget_fill_fmt(struct device *dev,
-		struct skl_module_cfg *mconfig,
+		struct skl_module_iface *fmt,
 		u32 tkn, u32 val, u32 dir, int fmt_idx)
 {
 	struct skl_module_fmt *dst_fmt;
 
+	if (!fmt)
+		return -EINVAL;
+
 	switch (dir) {
 	case SKL_DIR_IN:
-		dst_fmt = &mconfig->in_fmt[fmt_idx];
+		dst_fmt = &fmt->inputs[fmt_idx].fmt;
 		break;
 
 	case SKL_DIR_OUT:
-		dst_fmt = &mconfig->out_fmt[fmt_idx];
+		dst_fmt = &fmt->outputs[fmt_idx].fmt;
 		break;
 
 	default:
@@ -2117,6 +2272,10 @@ static int skl_tplg_fill_res_tkn(struct device *dev,
 		res->ibs = tkn_elem->value;
 		break;
 
+	case SKL_TKN_U32_MAX_MCPS:
+		res->cps = tkn_elem->value;
+		break;
+
 	case SKL_TKN_MM_U32_RES_PIN_ID:
 	case SKL_TKN_MM_U32_PIN_BUF:
 		ret = skl_tplg_manifest_pin_res_tkn(dev, tkn_elem, res,
@@ -2145,49 +2304,54 @@ static int skl_tplg_get_token(struct device *dev,
 	int tkn_count = 0;
 	int ret;
 	static int is_pipe_exists;
-	static int pin_index, dir;
+	static int pin_index, dir, conf_idx;
+	struct skl_module_iface *iface = NULL;
+	struct skl_module_res *res = NULL;
+	int res_idx = mconfig->res_idx;
+	int fmt_idx = mconfig->fmt_idx;
+
+	/*
+	 * If the manifest structure contains no modules, fill all
+	 * the module data to 0th index.
+	 * res_idx and fmt_idx are default set to 0.
+	 */
+	if (skl->nr_modules == 0) {
+		res = &mconfig->module->resources[res_idx];
+		iface = &mconfig->module->formats[fmt_idx];
+	}
 
 	if (tkn_elem->token > SKL_TKN_MAX)
 		return -EINVAL;
 
 	switch (tkn_elem->token) {
 	case SKL_TKN_U8_IN_QUEUE_COUNT:
-		mconfig->max_in_queue = tkn_elem->value;
-		mconfig->m_in_pin = devm_kzalloc(dev, mconfig->max_in_queue *
-					sizeof(*mconfig->m_in_pin),
-					GFP_KERNEL);
-		if (!mconfig->m_in_pin)
-			return -ENOMEM;
-
+		mconfig->module->max_input_pins = tkn_elem->value;
 		break;
 
 	case SKL_TKN_U8_OUT_QUEUE_COUNT:
-		mconfig->max_out_queue = tkn_elem->value;
-		mconfig->m_out_pin = devm_kzalloc(dev, mconfig->max_out_queue *
-					sizeof(*mconfig->m_out_pin),
-					GFP_KERNEL);
-
-		if (!mconfig->m_out_pin)
-			return -ENOMEM;
-
+		mconfig->module->max_output_pins = tkn_elem->value;
 		break;
 
 	case SKL_TKN_U8_DYN_IN_PIN:
+		if (!mconfig->m_in_pin)
+			mconfig->m_in_pin = devm_kzalloc(dev, MAX_IN_QUEUE *
+					sizeof(*mconfig->m_in_pin), GFP_KERNEL);
 		if (!mconfig->m_in_pin)
 			return -ENOMEM;
 
-		skl_tplg_fill_pin_dynamic_val(mconfig->m_in_pin,
-			mconfig->max_in_queue, tkn_elem->value);
-
+		skl_tplg_fill_pin_dynamic_val(mconfig->m_in_pin, MAX_IN_QUEUE,
+					      tkn_elem->value);
 		break;
 
 	case SKL_TKN_U8_DYN_OUT_PIN:
+		if (!mconfig->m_out_pin)
+			mconfig->m_out_pin = devm_kzalloc(dev, MAX_IN_QUEUE *
+					sizeof(*mconfig->m_in_pin), GFP_KERNEL);
 		if (!mconfig->m_out_pin)
 			return -ENOMEM;
 
-		skl_tplg_fill_pin_dynamic_val(mconfig->m_out_pin,
-			mconfig->max_out_queue, tkn_elem->value);
-
+		skl_tplg_fill_pin_dynamic_val(mconfig->m_out_pin, MAX_OUT_QUEUE,
+					      tkn_elem->value);
 		break;
 
 	case SKL_TKN_U8_TIME_SLOT:
@@ -2215,19 +2379,13 @@ static int skl_tplg_get_token(struct device *dev,
 		break;
 
 	case SKL_TKN_U32_MEM_PAGES:
-		mconfig->mem_pages = tkn_elem->value;
-		break;
-
 	case SKL_TKN_U32_MAX_MCPS:
-		mconfig->mcps = tkn_elem->value;
-		break;
-
 	case SKL_TKN_U32_OBS:
-		mconfig->obs = tkn_elem->value;
-		break;
-
 	case SKL_TKN_U32_IBS:
-		mconfig->ibs = tkn_elem->value;
+		ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, dir, pin_index);
+		if (ret < 0)
+			return ret;
+
 		break;
 
 	case SKL_TKN_U32_VBUS_ID:
@@ -2260,10 +2418,16 @@ static int skl_tplg_get_token(struct device *dev,
 
 		break;
 
+	case SKL_TKN_U32_PIPE_CONFIG_ID:
+		conf_idx = tkn_elem->value;
+		break;
+
 	case SKL_TKN_U32_PIPE_CONN_TYPE:
 	case SKL_TKN_U32_PIPE_PRIORITY:
 	case SKL_TKN_U32_PIPE_MEM_PGS:
 	case SKL_TKN_U32_PMODE:
+	case SKL_TKN_U32_PIPE_DIRECTION:
+	case SKL_TKN_U32_NUM_CONFIGS:
 		if (is_pipe_exists) {
 			ret = skl_tplg_fill_pipe_tkn(dev, mconfig->pipe,
 					tkn_elem->token, tkn_elem->value);
@@ -2273,6 +2437,27 @@ static int skl_tplg_get_token(struct device *dev,
 
 		break;
 
+	case SKL_TKN_U32_PATH_MEM_PGS:
+	case SKL_TKN_U32_CFG_FREQ:
+	case SKL_TKN_U8_CFG_CHAN:
+	case SKL_TKN_U8_CFG_BPS:
+		if (mconfig->pipe->nr_cfgs) {
+			ret = skl_tplg_fill_pipe_cfg(dev, mconfig->pipe,
+					tkn_elem->token, tkn_elem->value,
+					conf_idx, dir);
+			if (ret < 0)
+				return ret;
+		}
+		break;
+
+	case SKL_TKN_CFG_MOD_RES_ID:
+		mconfig->mod_cfg[conf_idx].res_idx = tkn_elem->value;
+		break;
+
+	case SKL_TKN_CFG_MOD_FMT_ID:
+		mconfig->mod_cfg[conf_idx].fmt_idx = tkn_elem->value;
+		break;
+
 	/*
 	 * SKL_TKN_U32_DIR_PIN_COUNT token has the value for both
 	 * direction and the pin count. The first four bits represent
@@ -2293,7 +2478,7 @@ static int skl_tplg_get_token(struct device *dev,
 	case SKL_TKN_U32_FMT_INTERLEAVE:
 	case SKL_TKN_U32_FMT_SAMPLE_TYPE:
 	case SKL_TKN_U32_FMT_CH_MAP:
-		ret = skl_tplg_widget_fill_fmt(dev, mconfig, tkn_elem->token,
+		ret = skl_tplg_widget_fill_fmt(dev, iface, tkn_elem->token,
 				tkn_elem->value, dir, pin_index);
 
 		if (ret < 0)
@@ -2518,11 +2703,11 @@ static void skl_clear_pin_config(struct snd_soc_platform *platform,
 					strlen(platform->component.name))) {
 		mconfig = w->priv;
 		pipe = mconfig->pipe;
-		for (i = 0; i < mconfig->max_in_queue; i++) {
+		for (i = 0; i < mconfig->module->max_input_pins; i++) {
 			mconfig->m_in_pin[i].in_use = false;
 			mconfig->m_in_pin[i].pin_state = SKL_PIN_UNBIND;
 		}
-		for (i = 0; i < mconfig->max_out_queue; i++) {
+		for (i = 0; i < mconfig->module->max_output_pins; i++) {
 			mconfig->m_out_pin[i].in_use = false;
 			mconfig->m_out_pin[i].pin_state = SKL_PIN_UNBIND;
 		}
@@ -2581,6 +2766,13 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
 	if (!mconfig)
 		return -ENOMEM;
 
+	if (skl->nr_modules == 0) {
+		mconfig->module = devm_kzalloc(bus->dev,
+				sizeof(*mconfig->module), GFP_KERNEL);
+		if (!mconfig->module)
+			return -ENOMEM;
+	}
+
 	w->priv = mconfig;
 
 	/*
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 93b2ba8c40f7..90903a15fa1d 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -273,6 +273,23 @@ struct skl_pipe_params {
 	unsigned int link_bps;
 };
 
+struct skl_pipe_fmt {
+	u32 freq;
+	u8 channels;
+	u8 bps;
+};
+
+struct skl_pipe_mcfg {
+	u8 res_idx;
+	u8 fmt_idx;
+};
+
+struct skl_path_config {
+	u8 mem_pages;
+	struct skl_pipe_fmt in_fmt;
+	struct skl_pipe_fmt out_fmt;
+};
+
 struct skl_pipe {
 	u8 ppl_id;
 	u8 pipe_priority;
@@ -281,6 +298,10 @@ struct skl_pipe {
 	u8 lp_mode;
 	struct skl_pipe_params *p_params;
 	enum skl_pipe_state state;
+	u8 direction;
+	u8 cur_config_idx;
+	u8 nr_cfgs;
+	struct skl_path_config configs[SKL_MAX_PATH_CONFIGS];
 	struct list_head w_list;
 	bool passthru;
 };
@@ -347,6 +368,9 @@ struct skl_module {
 struct skl_module_cfg {
 	u8 guid[16];
 	struct skl_module_inst_id id;
+	struct skl_module *module;
+	int res_idx;
+	int fmt_idx;
 	u8 domain;
 	bool homogenous_inputs;
 	bool homogenous_outputs;
@@ -381,6 +405,7 @@ struct skl_module_cfg {
 	enum skl_module_state m_state;
 	struct skl_pipe *pipe;
 	struct skl_specific_cfg formats_config;
+	struct skl_pipe_mcfg mod_cfg[SKL_MAX_MODULES_IN_PIPE];
 };
 
 struct skl_algo_data {
-- 
cgit v1.2.3


From 91fe0e70dbfb1142e17cebaa9186794b51a28cf7 Mon Sep 17 00:00:00 2001
From: Jeeja KP
Date: Wed, 23 Aug 2017 19:33:54 +0530
Subject: ASoC: Intel: Skylake: Update module id in pin connections

Each module's id comes from the topology and gets updated in the
driver. This patch updates the input and output pin connections of
each module by matching the uuid for each module.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-pcm.c      | 41 +++++++++++++++++++++++++++++++---
 sound/soc/intel/skylake/skl-topology.h |  1 +
 2 files changed, 39 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index a756882b9029..2b1e513b1680 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -1220,8 +1220,11 @@ static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd)
 static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig)
 {
 	struct skl_sst *ctx = skl->skl_sst;
+	struct skl_module_inst_id *pin_id;
+	uuid_le *uuid_mod, *uuid_tplg;
+	struct skl_module *skl_module;
 	struct uuid_module *module;
-	uuid_le *uuid_mod;
+	int i, ret = -EIO;
 
 	uuid_mod = (uuid_le *)mconfig->guid;
 
@@ -1235,11 +1238,43 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig)
 			mconfig->id.module_id = module->id;
 			if (mconfig->module)
 				mconfig->module->loadable = module->is_loadable;
-			return 0;
+			ret = 0;
+			break;
 		}
 	}
 
-	return -EIO;
+	if (ret)
+		return ret;
+
+	uuid_mod = &module->uuid;
+	ret = -EIO;
+	for (i = 0; i < skl->nr_modules; i++) {
+		skl_module = skl->modules[i];
+		uuid_tplg = &skl_module->uuid;
+		if (!uuid_le_cmp(*uuid_mod, *uuid_tplg)) {
+			mconfig->module = skl_module;
+			ret = 0;
+			break;
+		}
+	}
+	if (skl->nr_modules && ret)
+		return ret;
+
+	list_for_each_entry(module, &ctx->uuid_list, list) {
+		for (i = 0; i < MAX_IN_QUEUE; i++) {
+			pin_id = &mconfig->m_in_pin[i].id;
+			if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid))
+				pin_id->module_id = module->id;
+		}
+
+		for (i = 0; i < MAX_OUT_QUEUE; i++) {
+			pin_id = &mconfig->m_out_pin[i].id;
+			if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid))
+				pin_id->module_id = module->id;
+		}
+	}
+
+	return 0;
 }
 
 static int skl_populate_modules(struct skl *skl)
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 90903a15fa1d..4975904480e9 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -221,6 +221,7 @@ struct skl_kpb_params {
 };
 
 struct skl_module_inst_id {
+	uuid_le mod_uuid;
 	int module_id;
 	u32 instance_id;
 	int pvt_id;
-- 
cgit v1.2.3


From 88c27465dc7e3f11c43de295f623c44f593a0912 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Fri, 25 Aug 2017 01:06:04 +0000
Subject: ASoC: use snd_soc_component_get_dapm()

Now we have snd_soc_component_get_dapm(),
and as soc.h say below, let's use it.

/* Don't use these, use snd_soc_component_get_dapm() */
struct snd_soc_dapm_context dapm;

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 74c17068bb11..92e09cb7fda4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3220,7 +3220,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	component->pcm_new = component->driver->pcm_new;
 	component->pcm_free = component->driver->pcm_free;
 
-	dapm = &component->dapm;
+	dapm = snd_soc_component_get_dapm(component);
 	dapm->dev = dev;
 	dapm->component = component;
 	dapm->bias_level = SND_SOC_BIAS_OFF;
-- 
cgit v1.2.3


From 3dfcd3468cf05861654b1c7857500da6c8a9fc88 Mon Sep 17 00:00:00 2001
From: John Hsu
Date: Fri, 25 Aug 2017 11:35:44 +0800
Subject: ASoC: nau8825: correct typo of semaphore comment

There are a lot of typo about semaphore in the comment.
Correct it from semaphone to semaphore.

Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/nau8825.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 46a30eaa7ace..a97b0b6d9086 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -260,11 +260,11 @@ static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout)
 	if (timeout) {
 		ret = down_timeout(&nau8825->xtalk_sem, timeout);
 		if (ret < 0)
-			dev_warn(nau8825->dev, "Acquire semaphone timeout\n");
+			dev_warn(nau8825->dev, "Acquire semaphore timeout\n");
 	} else {
 		ret = down_interruptible(&nau8825->xtalk_sem);
 		if (ret < 0)
-			dev_warn(nau8825->dev, "Acquire semaphone fail\n");
+			dev_warn(nau8825->dev, "Acquire semaphore fail\n");
 	}
 
 	return ret;
@@ -1299,7 +1299,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream,
 	regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1,
 		NAU8825_I2S_DL_MASK, val_len);
 
-	/* Release the semaphone. */
+	/* Release the semaphore. */
 	nau8825_sema_release(nau8825);
 
 	return 0;
@@ -1361,7 +1361,7 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2,
 		NAU8825_I2S_MS_MASK, ctrl2_val);
 
-	/* Release the semaphone. */
+	/* Release the semaphore. */
 	nau8825_sema_release(nau8825);
 
 	return 0;
@@ -2140,7 +2140,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 
 		break;
 	case NAU8825_CLK_MCLK:
-		/* Acquire the semaphone to synchronize the playback and
+		/* Acquire the semaphore to synchronize the playback and
 		 * interrupt handler. In order to avoid the playback inter-
 		 * fered by cross talk process, the driver make the playback
 		 * preparation halted until cross talk process finish.
@@ -2150,7 +2150,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 		/* MCLK not changed by clock tree */
 		regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER,
 			NAU8825_CLK_MCLK_SRC_MASK, 0);
-		/* Release the semaphone. */
+		/* Release the semaphore. */
 		nau8825_sema_release(nau8825);
 
 		ret = nau8825_mclk_prepare(nau8825, freq);
@@ -2188,7 +2188,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 
 		break;
 	case NAU8825_CLK_FLL_MCLK:
-		/* Acquire the semaphone to synchronize the playback and
+		/* Acquire the semaphore to synchronize the playback and
 		 * interrupt handler. In order to avoid the playback inter-
 		 * fered by cross talk process, the driver make the playback
 		 * preparation halted until cross talk process finish.
@@ -2201,7 +2201,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 		regmap_update_bits(regmap, NAU8825_REG_FLL3,
 			NAU8825_FLL_CLK_SRC_MASK | NAU8825_GAIN_ERR_MASK,
 			NAU8825_FLL_CLK_SRC_MCLK | 0);
-		/* Release the semaphone. */
+		/* Release the semaphore. */
 		nau8825_sema_release(nau8825);
 
 		ret = nau8825_mclk_prepare(nau8825, freq);
@@ -2210,7 +2210,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 
 		break;
 	case NAU8825_CLK_FLL_BLK:
-		/* Acquire the semaphone to synchronize the playback and
+		/* Acquire the semaphore to synchronize the playback and
 		 * interrupt handler. In order to avoid the playback inter-
 		 * fered by cross talk process, the driver make the playback
 		 * preparation halted until cross talk process finish.
@@ -2226,7 +2226,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 			NAU8825_FLL_CLK_SRC_MASK | NAU8825_GAIN_ERR_MASK,
 			NAU8825_FLL_CLK_SRC_BLK |
 			(0xf << NAU8825_GAIN_ERR_SFT));
-		/* Release the semaphone. */
+		/* Release the semaphore. */
 		nau8825_sema_release(nau8825);
 
 		if (nau8825->mclk_freq) {
@@ -2236,7 +2236,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 
 		break;
 	case NAU8825_CLK_FLL_FS:
-		/* Acquire the semaphone to synchronize the playback and
+		/* Acquire the semaphore to synchronize the playback and
 		 * interrupt handler. In order to avoid the playback inter-
 		 * fered by cross talk process, the driver make the playback
 		 * preparation halted until cross talk process finish.
@@ -2252,7 +2252,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
 			NAU8825_FLL_CLK_SRC_MASK | NAU8825_GAIN_ERR_MASK,
 			NAU8825_FLL_CLK_SRC_FS |
 			(0xf << NAU8825_GAIN_ERR_SFT));
-		/* Release the semaphone. */
+		/* Release the semaphore. */
 		nau8825_sema_release(nau8825);
 
 		if (nau8825->mclk_freq) {
@@ -2563,7 +2563,7 @@ static int nau8825_i2c_probe(struct i2c_client *i2c,
 		return PTR_ERR(nau8825->regmap);
 	nau8825->dev = dev;
 	nau8825->irq = i2c->irq;
-	/* Initiate parameters, semaphone and work queue which are needed in
+	/* Initiate parameters, semaphore and work queue which are needed in
 	 * cross talk suppression measurment function.
 	 */
 	nau8825->xtalk_state = NAU8825_XTALK_DONE;
-- 
cgit v1.2.3


From b8972bf0521ca7ee3c8d27da0fd101fe648acfc2 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Fri, 25 Aug 2017 00:29:01 +0000
Subject: ASoC: remove duplicate definition of controls/num_controls

snd_soc_component and snd_soc_component_driver both have
controls/num_controls, but these are duplicated.
Let's remove duplicated definition.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 2 --
 sound/soc/soc-core.c | 9 ++++-----
 2 files changed, 4 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index de57a8ad057b..e4b4746032ba 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -862,8 +862,6 @@ struct snd_soc_component {
 	/* Don't use these, use snd_soc_component_get_dapm() */
 	struct snd_soc_dapm_context dapm;
 
-	const struct snd_kcontrol_new *controls;
-	unsigned int num_controls;
 	const struct snd_soc_dapm_widget *dapm_widgets;
 	unsigned int num_dapm_widgets;
 	const struct snd_soc_dapm_route *dapm_routes;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3bb8c63564cf..1317706f8e69 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1495,9 +1495,10 @@ static int soc_probe_component(struct snd_soc_card *card,
 		}
 	}
 
-	if (component->controls)
-		snd_soc_add_component_controls(component, component->controls,
-				     component->num_controls);
+	if (component->driver->controls)
+		snd_soc_add_component_controls(component,
+					       component->driver->controls,
+					       component->driver->num_controls);
 	if (component->dapm_routes)
 		snd_soc_dapm_add_routes(dapm, component->dapm_routes,
 					component->num_dapm_routes);
@@ -3184,8 +3185,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	if (driver->stream_event)
 		dapm->stream_event = snd_soc_component_stream_event;
 
-	component->controls = driver->controls;
-	component->num_controls = driver->num_controls;
 	component->dapm_widgets = driver->dapm_widgets;
 	component->num_dapm_widgets = driver->num_dapm_widgets;
 	component->dapm_routes = driver->dapm_routes;
-- 
cgit v1.2.3


From 688d0ebf49a97aa6959aebec5d177a8b821f27d1 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Fri, 25 Aug 2017 00:29:20 +0000
Subject: ASoC: remove duplicate definition of dapm_widgets/num_dapm_widgets

snd_soc_component and snd_soc_component_driver both have
dapm_widgets/num_dapm_widgets, but these are duplicated.
Let's remove duplicated definition.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 2 --
 sound/soc/soc-core.c | 9 ++++-----
 2 files changed, 4 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index e4b4746032ba..3e0070ed6883 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -862,8 +862,6 @@ struct snd_soc_component {
 	/* Don't use these, use snd_soc_component_get_dapm() */
 	struct snd_soc_dapm_context dapm;
 
-	const struct snd_soc_dapm_widget *dapm_widgets;
-	unsigned int num_dapm_widgets;
 	const struct snd_soc_dapm_route *dapm_routes;
 	unsigned int num_dapm_routes;
 	struct snd_soc_codec *codec;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1317706f8e69..3a3236dc996d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1451,9 +1451,10 @@ static int soc_probe_component(struct snd_soc_card *card,
 
 	soc_init_component_debugfs(component);
 
-	if (component->dapm_widgets) {
-		ret = snd_soc_dapm_new_controls(dapm, component->dapm_widgets,
-			component->num_dapm_widgets);
+	if (component->driver->dapm_widgets) {
+		ret = snd_soc_dapm_new_controls(dapm,
+					component->driver->dapm_widgets,
+					component->driver->num_dapm_widgets);
 
 		if (ret != 0) {
 			dev_err(component->dev,
@@ -3185,8 +3186,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	if (driver->stream_event)
 		dapm->stream_event = snd_soc_component_stream_event;
 
-	component->dapm_widgets = driver->dapm_widgets;
-	component->num_dapm_widgets = driver->num_dapm_widgets;
 	component->dapm_routes = driver->dapm_routes;
 	component->num_dapm_routes = driver->num_dapm_routes;
 
-- 
cgit v1.2.3


From 6969b2bae6dbe7f277a7c018b66d2f3ad662129f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Fri, 25 Aug 2017 00:29:41 +0000
Subject: ASoC: remove duplicate definition of dapm_routes/num_dapm_routes

snd_soc_component and snd_soc_component_driver both have
dapm_routes/num_dapm_routes, but these are duplicated.
Let's remove duplicated definition.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  2 --
 sound/soc/soc-core.c | 10 ++++------
 2 files changed, 4 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3e0070ed6883..2bcb38045dc1 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -862,8 +862,6 @@ struct snd_soc_component {
 	/* Don't use these, use snd_soc_component_get_dapm() */
 	struct snd_soc_dapm_context dapm;
 
-	const struct snd_soc_dapm_route *dapm_routes;
-	unsigned int num_dapm_routes;
 	struct snd_soc_codec *codec;
 
 	int (*probe)(struct snd_soc_component *);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3a3236dc996d..b8f96a41b594 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1500,9 +1500,10 @@ static int soc_probe_component(struct snd_soc_card *card,
 		snd_soc_add_component_controls(component,
 					       component->driver->controls,
 					       component->driver->num_controls);
-	if (component->dapm_routes)
-		snd_soc_dapm_add_routes(dapm, component->dapm_routes,
-					component->num_dapm_routes);
+	if (component->driver->dapm_routes)
+		snd_soc_dapm_add_routes(dapm,
+					component->driver->dapm_routes,
+					component->driver->num_dapm_routes);
 
 	list_add(&dapm->list, &card->dapm_list);
 	list_add(&component->card_list, &card->component_dev_list);
@@ -3186,9 +3187,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	if (driver->stream_event)
 		dapm->stream_event = snd_soc_component_stream_event;
 
-	component->dapm_routes = driver->dapm_routes;
-	component->num_dapm_routes = driver->num_dapm_routes;
-
 	INIT_LIST_HEAD(&component->dai_list);
 	mutex_init(&component->io_mutex);
 
-- 
cgit v1.2.3


From 4243e0457c9b1fe5e96fba19296469466eb24b1d Mon Sep 17 00:00:00 2001
From: Christophe Jaillet
Date: Sun, 27 Aug 2017 08:46:50 +0200
Subject: ASoC: davinci-mcasp: check memory allocation failure

Check memory allocation failures and return -ENOMEM in such cases, as
already done above for another memory allocation.

This avoids NULL pointers dereference.

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index d486908d4e0e..f395bbc7c354 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1851,6 +1851,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev,
 					sizeof(u32) * mcasp->num_serializer,
 					GFP_KERNEL);
+	if (!mcasp->context.xrsr_regs) {
+		ret = -ENOMEM;
+		goto err;
+	}
 #endif
 	mcasp->serial_dir = pdata->serial_dir;
 	mcasp->version = pdata->version;
-- 
cgit v1.2.3


From c20252cf9230eb7abd8bce9fdb2ce9bdd6743689 Mon Sep 17 00:00:00 2001
From: Naveen M
Date: Sat, 26 Aug 2017 04:04:09 +0530
Subject: ASoC: Intel: kbl: Add MST route change to kbl machine drivers

To support MST hdmi audio, modify the current routes to be based
on port in rt5663_max98927 & rt5663_rt5514_max98927 machine.

Signed-off-by: Naveen M <naveen.m@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c        | 8 +++-----
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 7 ++-----
 2 files changed, 5 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index c1442147d138..941480c3e906 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -74,8 +74,9 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = {
 	SND_SOC_DAPM_SPK("Left Spk", NULL),
 	SND_SOC_DAPM_SPK("Right Spk", NULL),
 	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
-	SND_SOC_DAPM_SPK("DP", NULL),
-	SND_SOC_DAPM_SPK("HDMI", NULL),
+	SND_SOC_DAPM_SPK("HDMI1", NULL),
+	SND_SOC_DAPM_SPK("HDMI2", NULL),
+	SND_SOC_DAPM_SPK("HDMI3", NULL),
 
 };
 
@@ -93,9 +94,6 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 	{ "IN1N", NULL, "Headset Mic" },
 	{ "DMic", NULL, "SoC DMIC" },
 
-	{ "HDMI", NULL, "hif5 Output" },
-	{ "DP", NULL, "hif6 Output" },
-
 	/* CODEC BE connections */
 	{ "Left HiFi Playback", NULL, "ssp0 Tx" },
 	{ "Right HiFi Playback", NULL, "ssp0 Tx" },
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 69681b04d7fc..d042255235c1 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -85,8 +85,8 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = {
 	SND_SOC_DAPM_SPK("Left Spk", NULL),
 	SND_SOC_DAPM_SPK("Right Spk", NULL),
 	SND_SOC_DAPM_MIC("DMIC", NULL),
-	SND_SOC_DAPM_SPK("DP", NULL),
-	SND_SOC_DAPM_SPK("HDMI", NULL),
+	SND_SOC_DAPM_SPK("HDMI1", NULL),
+	SND_SOC_DAPM_SPK("HDMI2", NULL),
 
 };
 
@@ -103,9 +103,6 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 	{ "IN1P", NULL, "Headset Mic" },
 	{ "IN1N", NULL, "Headset Mic" },
 
-	{ "HDMI", NULL, "hif5 Output" },
-	{ "DP", NULL, "hif6 Output" },
-
 	/* CODEC BE connections */
 	{ "Left HiFi Playback", NULL, "ssp0 Tx" },
 	{ "Right HiFi Playback", NULL, "ssp0 Tx" },
-- 
cgit v1.2.3


From 3a13347f05fdba9d889891245f154816197868f4 Mon Sep 17 00:00:00 2001
From: Naveen M
Date: Sat, 26 Aug 2017 04:04:10 +0530
Subject: ASoC: Intel: kbl: Add jack port initialize in kbl machine drivers

After the pcm jack is created, create and initialize the pin switch
widget for each port. For hdmi audio, invoke hdac_hdmi_jack_port_init
func() in rt5663_max98927 & rt5663_rt5514_max98927 to enable the pin,
when monitor is connected.

Signed-off-by: Naveen M <naveen.m@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c        | 7 ++++++-
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 7 ++++++-
 2 files changed, 12 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 941480c3e906..3fa85c32b9f8 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -767,10 +767,12 @@ static int kabylake_card_late_probe(struct snd_soc_card *card)
 {
 	struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(card);
 	struct kbl_hdmi_pcm *pcm;
+	struct snd_soc_codec *codec = NULL;
 	int err, i = 0;
 	char jack_name[NAME_SIZE];
 
 	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		codec = pcm->codec_dai->codec;
 		snprintf(jack_name, sizeof(jack_name),
 			"HDMI/DP, pcm=%d Jack", pcm->device);
 		err = snd_soc_card_jack_new(card, jack_name,
@@ -788,7 +790,10 @@ static int kabylake_card_late_probe(struct snd_soc_card *card)
 		i++;
 	}
 
-	return 0;
+	if (!codec)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(codec, &card->dapm);
 }
 
 /* kabylake audio machine driver for SPT + RT5663 */
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index d042255235c1..11bd5ef1e3b2 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -600,10 +600,12 @@ static int kabylake_card_late_probe(struct snd_soc_card *card)
 {
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card);
 	struct kbl_hdmi_pcm *pcm;
+	struct snd_soc_codec *codec = NULL;
 	int err, i = 0;
 	char jack_name[NAME_SIZE];
 
 	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		codec = pcm->codec_dai->codec;
 		err = snd_soc_card_jack_new(card, jack_name,
 				SND_JACK_AVOUT, &ctx->kabylake_hdmi[i],
 				NULL, 0);
@@ -617,7 +619,10 @@ static int kabylake_card_late_probe(struct snd_soc_card *card)
 		i++;
 	}
 
-	return 0;
+	if (!codec)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(codec, &card->dapm);
 }
 
 /*
-- 
cgit v1.2.3


From b1dc00abcf18d27b36de500c150be88022c82270 Mon Sep 17 00:00:00 2001
From: Rob Herring
Date: Tue, 29 Aug 2017 07:37:55 -0500
Subject: ASoC: fsl_dma: remove dma_object path member

dma_object.path is unused, so rather than fix it to work with DT
full_name changes, just remove it.

Signed-off-by: Rob Herring <robh@kernel.org>
Cc: Timur Tabi <timur@tabi.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <fabio.estevam@nxp.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_dma.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index c17359648915..0c11f434a374 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -63,7 +63,6 @@ struct dma_object {
 	struct ccsr_dma_channel __iomem *channel;
 	unsigned int irq;
 	bool assigned;
-	char path[1];
 };
 
 /*
@@ -903,13 +902,12 @@ static int fsl_soc_dma_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
+	dma = kzalloc(sizeof(*dma), GFP_KERNEL);
 	if (!dma) {
 		of_node_put(ssi_np);
 		return -ENOMEM;
 	}
 
-	strcpy(dma->path, np->full_name);
 	dma->dai.ops = &fsl_dma_ops;
 	dma->dai.pcm_new = fsl_dma_new;
 	dma->dai.pcm_free = fsl_dma_free_dma_buffers;
-- 
cgit v1.2.3


From eb33869c720656c16e614ea40a175a014f60007b Mon Sep 17 00:00:00 2001
From: Hsin-Yu Chao
Date: Mon, 28 Aug 2017 15:13:50 +0800
Subject: ASoC: rt5514: Guard Hotword Model bytes loading

If the rt5514-spi driver is not enabled in kernel, hotword model will
not be loaded when "DSP Voice Wake Up" is set to turn on DSP mode, and
an error is logged instead.

Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5514.c | 13 ++++++++++++-
 1 file changed, 12 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 0e239e8c7bf8..23fccdf3e84a 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -302,7 +302,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 	struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component);
 	struct snd_soc_codec *codec = rt5514->codec;
 	const struct firmware *fw = NULL;
-	int ret = 0;
 
 	if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled)
 		return 0;
@@ -340,6 +339,9 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 			}
 
 			if (rt5514->model_buf && rt5514->model_len) {
+#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI)
+				int ret;
+
 				ret = rt5514_spi_burst_write(0x4ff80000,
 					rt5514->model_buf,
 					((rt5514->model_len / 8) + 1) * 8);
@@ -348,13 +350,22 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol,
 						"Model load failed %d\n", ret);
 					return ret;
 				}
+#else
+				dev_err(codec->dev,
+					"No SPI driver for loading firmware\n");
+#endif
 			} else {
 				request_firmware(&fw, RT5514_FIRMWARE3,
 						 codec->dev);
 				if (fw) {
+#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI)
 					rt5514_spi_burst_write(0x4ff80000,
 						fw->data,
 						((fw->size/8)+1)*8);
+#else
+					dev_err(codec->dev,
+						"No SPI driver to load fw\n");
+#endif
 					release_firmware(fw);
 					fw = NULL;
 				}
-- 
cgit v1.2.3


From 64220b9d10a5546b6dd1154ccaf43522ae7d02fc Mon Sep 17 00:00:00 2001
From: Naveen Manohar
Date: Sat, 26 Aug 2017 03:21:39 +0530
Subject: ASoC: Intel: kbl: Add map for Maxim IV Feedback

MAX98927 provides IV feedback on the capture widget.
Here we are connecting the capture widget to SSP0_RX and
SSP0_RX to the algorithm running on host.

Signed-off-by: Naveen M <naveen.m@intel.com>
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_max98927.c        | 5 +++++
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +++++
 2 files changed, 10 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 3fa85c32b9f8..7f7607420706 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -105,6 +105,11 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 	{ "hs_in", NULL, "ssp1 Rx" },
 	{ "ssp1 Rx", NULL, "AIF Capture" },
 
+	/* IV feedback path */
+	{ "codec0_fb_in", NULL, "ssp0 Rx"},
+	{ "ssp0 Rx", NULL, "Left HiFi Capture" },
+	{ "ssp0 Rx", NULL, "Right HiFi Capture" },
+
 	/* DMIC */
 	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
 	{ "DMIC01 Rx", NULL, "DMIC AIF" },
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 11bd5ef1e3b2..4689991bcc86 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -117,6 +117,11 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
 	{ "codec1_in", NULL, "ssp0 Rx" },
 	{ "ssp0 Rx", NULL, "AIF1 Capture" },
 
+	/* IV feedback path */
+	{ "codec0_fb_in", NULL, "ssp0 Rx"},
+	{ "ssp0 Rx", NULL, "Left HiFi Capture" },
+	{ "ssp0 Rx", NULL, "Right HiFi Capture" },
+
 	/* DMIC */
 	{ "DMIC1L", NULL, "DMIC" },
 	{ "DMIC1R", NULL, "DMIC" },
-- 
cgit v1.2.3


From 72cfb0f20c75c1e580597c164b9d74426e583f37 Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Thu, 24 Aug 2017 12:52:21 +0800
Subject: ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp

Currently we are using codec name for rt5514 dsp dai link, use codec
of_node instead.

Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Reviewed-by: Matthias Kaehlcke <mka@chromium.org>
Tested-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rk3399_gru_sound.c | 34 ++--------------------------------
 1 file changed, 2 insertions(+), 32 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index 566ccb39fb31..cd2fdba922f1 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -247,8 +247,6 @@ enum {
 	DAILINK_RT5514_DSP,
 };
 
-#define DAILINK_ENTITIES	(DAILINK_DA7219 + 1)
-
 static struct snd_soc_dai_link rockchip_dailinks[] = {
 	[DAILINK_MAX98357A] = {
 		.name = "MAX98357A",
@@ -282,8 +280,7 @@ static struct snd_soc_dai_link rockchip_dailinks[] = {
 	[DAILINK_RT5514_DSP] = {
 		.name = "RT5514 DSP",
 		.stream_name = "Wake on Voice",
-		.codec_name = "snd-soc-dummy",
-		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_dai_name = "rt5514-dsp-cpu-dai",
 	},
 };
 
@@ -300,17 +297,10 @@ static struct snd_soc_card rockchip_sound_card = {
 	.num_controls = ARRAY_SIZE(rockchip_controls),
 };
 
-static int rockchip_sound_match_stub(struct device *dev, void *data)
-{
-	return 1;
-}
-
 static int rockchip_sound_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &rockchip_sound_card;
 	struct device_node *cpu_node;
-	struct device *dev;
-	struct device_driver *drv;
 	int i, ret;
 
 	cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0);
@@ -319,7 +309,7 @@ static int rockchip_sound_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	for (i = 0; i < DAILINK_ENTITIES; i++) {
+	for (i = 0; i < ARRAY_SIZE(rockchip_dailinks); i++) {
 		rockchip_dailinks[i].platform_of_node = cpu_node;
 		rockchip_dailinks[i].cpu_of_node = cpu_node;
 
@@ -332,22 +322,6 @@ static int rockchip_sound_probe(struct platform_device *pdev)
 		}
 	}
 
-	/**
-	 * To acquire the spi driver of the rt5514 and set the dai-links names
-	 * for soc_bind_dai_link
-	 */
-	drv = driver_find("rt5514", &spi_bus_type);
-	if (!drv) {
-		dev_err(&pdev->dev, "Can not find the rt5514 driver at the spi bus\n");
-		return -EINVAL;
-	}
-
-	dev = driver_find_device(drv, NULL, NULL, rockchip_sound_match_stub);
-	if (!dev) {
-		dev_err(&pdev->dev, "Can not find the rt5514 device\n");
-		return -ENODEV;
-	}
-
 	/* Set DMIC wakeup delay */
 	ret = device_property_read_u32(&pdev->dev, "dmic-wakeup-delay-ms",
 					&dmic_wakeup_delay);
@@ -357,10 +331,6 @@ static int rockchip_sound_probe(struct platform_device *pdev)
 			"no optional property 'dmic-wakeup-delay-ms' found, default: no delay\n");
 	}
 
-	rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL);
-	rockchip_dailinks[DAILINK_RT5514_DSP].cpu_dai_name = kstrdup_const(dev_name(dev), GFP_KERNEL);
-	rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL);
-
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-- 
cgit v1.2.3


From 0d52954ffe80b051ab79a105bd801485c4bbe20a Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Thu, 24 Aug 2017 12:52:24 +0800
Subject: ASoC: rockchip: Parse dai links from dts

Refactor rockchip_sound_probe, parse dai links from dts instead of
hard coding them.

Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Reviewed-by: Matthias Kaehlcke <mka@chromium.org>
Tested-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rk3399_gru_sound.c | 140 ++++++++++++++++++++++------------
 1 file changed, 91 insertions(+), 49 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index cd2fdba922f1..347c34d0db4c 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -240,14 +240,42 @@ static const struct snd_soc_ops rockchip_sound_da7219_ops = {
 	.hw_params = rockchip_sound_da7219_hw_params,
 };
 
+static struct snd_soc_card rockchip_sound_card = {
+	.name = "rk3399-gru-sound",
+	.owner = THIS_MODULE,
+	.dapm_widgets = rockchip_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets),
+	.dapm_routes = rockchip_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes),
+	.controls = rockchip_controls,
+	.num_controls = ARRAY_SIZE(rockchip_controls),
+};
+
 enum {
+	DAILINK_DA7219,
 	DAILINK_MAX98357A,
 	DAILINK_RT5514,
-	DAILINK_DA7219,
 	DAILINK_RT5514_DSP,
 };
 
-static struct snd_soc_dai_link rockchip_dailinks[] = {
+static const char * const dailink_compat[] = {
+	[DAILINK_DA7219] = "dlg,da7219",
+	[DAILINK_MAX98357A] = "maxim,max98357a",
+	[DAILINK_RT5514] = "realtek,rt5514-i2c",
+	[DAILINK_RT5514_DSP] = "realtek,rt5514-spi",
+};
+
+static const struct snd_soc_dai_link rockchip_dais[] = {
+	[DAILINK_DA7219] = {
+		.name = "DA7219",
+		.stream_name = "DA7219 PCM",
+		.codec_dai_name = "da7219-hifi",
+		.init = rockchip_sound_da7219_init,
+		.ops = &rockchip_sound_da7219_ops,
+		/* set da7219 as slave */
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+	},
 	[DAILINK_MAX98357A] = {
 		.name = "MAX98357A",
 		.stream_name = "MAX98357A PCM",
@@ -266,16 +294,6 @@ static struct snd_soc_dai_link rockchip_dailinks[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
 	},
-	[DAILINK_DA7219] = {
-		.name = "DA7219",
-		.stream_name = "DA7219 PCM",
-		.codec_dai_name = "da7219-hifi",
-		.init = rockchip_sound_da7219_init,
-		.ops = &rockchip_sound_da7219_ops,
-		/* set da7219 as slave */
-		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-			SND_SOC_DAIFMT_CBS_CFS,
-	},
 	/* RT5514 DSP for voice wakeup via spi bus */
 	[DAILINK_RT5514_DSP] = {
 		.name = "RT5514 DSP",
@@ -284,42 +302,72 @@ static struct snd_soc_dai_link rockchip_dailinks[] = {
 	},
 };
 
-static struct snd_soc_card rockchip_sound_card = {
-	.name = "rk3399-gru-sound",
-	.owner = THIS_MODULE,
-	.dai_link = rockchip_dailinks,
-	.num_links =  ARRAY_SIZE(rockchip_dailinks),
-	.dapm_widgets = rockchip_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets),
-	.dapm_routes = rockchip_dapm_routes,
-	.num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes),
-	.controls = rockchip_controls,
-	.num_controls = ARRAY_SIZE(rockchip_controls),
-};
-
-static int rockchip_sound_probe(struct platform_device *pdev)
+static int rockchip_sound_codec_node_match(struct device_node *np_codec)
 {
-	struct snd_soc_card *card = &rockchip_sound_card;
-	struct device_node *cpu_node;
-	int i, ret;
+	int i;
 
-	cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0);
-	if (!cpu_node) {
-		dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n");
-		return -EINVAL;
+	for (i = 0; i < ARRAY_SIZE(dailink_compat); i++) {
+		if (of_device_is_compatible(np_codec, dailink_compat[i]))
+			return i;
 	}
+	return -1;
+}
 
-	for (i = 0; i < ARRAY_SIZE(rockchip_dailinks); i++) {
-		rockchip_dailinks[i].platform_of_node = cpu_node;
-		rockchip_dailinks[i].cpu_of_node = cpu_node;
-
-		rockchip_dailinks[i].codec_of_node =
-			of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i);
-		if (!rockchip_dailinks[i].codec_of_node) {
-			dev_err(&pdev->dev,
-				"Property[%d] 'rockchip,codec' missing or invalid\n", i);
+static int rockchip_sound_of_parse_dais(struct device *dev,
+					struct snd_soc_card *card)
+{
+	struct device_node *np_cpu;
+	struct device_node *np_codec;
+	struct snd_soc_dai_link *dai;
+	int i, index;
+
+	card->dai_link = devm_kzalloc(dev, sizeof(rockchip_dais),
+				      GFP_KERNEL);
+	if (!card->dai_link)
+		return -ENOMEM;
+
+	np_cpu = of_parse_phandle(dev->of_node, "rockchip,cpu", 0);
+
+	card->num_links = 0;
+	for (i = 0; i < ARRAY_SIZE(rockchip_dais); i++) {
+		np_codec = of_parse_phandle(dev->of_node,
+					    "rockchip,codec", i);
+		if (!np_codec)
+			break;
+
+		if (!of_device_is_available(np_codec))
+			continue;
+
+		index = rockchip_sound_codec_node_match(np_codec);
+		if (index < 0)
+			continue;
+
+		if (!np_cpu) {
+			dev_err(dev, "Missing 'rockchip,cpu' for %s\n",
+				rockchip_dais[index].name);
 			return -EINVAL;
 		}
+
+		dai = &card->dai_link[card->num_links++];
+		*dai = rockchip_dais[index];
+
+		dai->codec_of_node = np_codec;
+		dai->platform_of_node = np_cpu;
+		dai->cpu_of_node = np_cpu;
+	}
+
+	return 0;
+}
+
+static int rockchip_sound_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &rockchip_sound_card;
+	int ret;
+
+	ret = rockchip_sound_of_parse_dais(&pdev->dev, card);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to parse dais: %d\n", ret);
+		return ret;
 	}
 
 	/* Set DMIC wakeup delay */
@@ -332,13 +380,7 @@ static int rockchip_sound_probe(struct platform_device *pdev)
 	}
 
 	card->dev = &pdev->dev;
-
-	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret)
-		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
-			__func__, ret);
-
-	return ret;
+	return devm_snd_soc_register_card(&pdev->dev, card);
 }
 
 static const struct of_device_id rockchip_sound_of_match[] = {
-- 
cgit v1.2.3


From 3313faf1053ee6cb90045f6877f11e2f61ead63a Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Thu, 24 Aug 2017 12:52:25 +0800
Subject: ASoC: rockchip: Add support for DP codec

Add support for optional cdn dp codec.

Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/Kconfig            |  1 +
 sound/soc/rockchip/rk3399_gru_sound.c | 59 +++++++++++++++++++++++++++++++++--
 2 files changed, 58 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
index c84487805876..8f0d0d8d34e6 100644
--- a/sound/soc/rockchip/Kconfig
+++ b/sound/soc/rockchip/Kconfig
@@ -68,6 +68,7 @@ config SND_SOC_RK3399_GRU_SOUND
 	select SND_SOC_RT5514
 	select SND_SOC_DA7219
 	select SND_SOC_RT5514_SPI
+	select SND_SOC_HDMI_CODEC
 	help
 	  Say Y or M here if you want to add support multiple codecs for SoC
 	  audio on Rockchip RK3399 GRU boards.
diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index 347c34d0db4c..91aab5c18f3b 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -228,6 +228,45 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
+static int rockchip_sound_cdndp_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_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int mclk, ret;
+
+	/* in bypass mode, the mclk has to be one of the frequencies below */
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 24000:
+	case 32000:
+	case 48000:
+	case 64000:
+	case 96000:
+		mclk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+		mclk = 11289600;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk,
+				     SND_SOC_CLOCK_OUT);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_ops rockchip_sound_max98357a_ops = {
 	.hw_params = rockchip_sound_max98357a_hw_params,
 };
@@ -240,6 +279,10 @@ static const struct snd_soc_ops rockchip_sound_da7219_ops = {
 	.hw_params = rockchip_sound_da7219_hw_params,
 };
 
+static struct snd_soc_ops rockchip_sound_cdndp_ops = {
+	.hw_params = rockchip_sound_cdndp_hw_params,
+};
+
 static struct snd_soc_card rockchip_sound_card = {
 	.name = "rk3399-gru-sound",
 	.owner = THIS_MODULE,
@@ -252,6 +295,7 @@ static struct snd_soc_card rockchip_sound_card = {
 };
 
 enum {
+	DAILINK_CDNDP,
 	DAILINK_DA7219,
 	DAILINK_MAX98357A,
 	DAILINK_RT5514,
@@ -259,6 +303,7 @@ enum {
 };
 
 static const char * const dailink_compat[] = {
+	[DAILINK_CDNDP] = "rockchip,rk3399-cdn-dp",
 	[DAILINK_DA7219] = "dlg,da7219",
 	[DAILINK_MAX98357A] = "maxim,max98357a",
 	[DAILINK_RT5514] = "realtek,rt5514-i2c",
@@ -266,6 +311,14 @@ static const char * const dailink_compat[] = {
 };
 
 static const struct snd_soc_dai_link rockchip_dais[] = {
+	[DAILINK_CDNDP] = {
+		.name = "DP",
+		.stream_name = "DP PCM",
+		.codec_dai_name = "i2s-hifi",
+		.ops = &rockchip_sound_cdndp_ops,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+	},
 	[DAILINK_DA7219] = {
 		.name = "DA7219",
 		.stream_name = "DA7219 PCM",
@@ -316,7 +369,7 @@ static int rockchip_sound_codec_node_match(struct device_node *np_codec)
 static int rockchip_sound_of_parse_dais(struct device *dev,
 					struct snd_soc_card *card)
 {
-	struct device_node *np_cpu;
+	struct device_node *np_cpu, *np_cpu0, *np_cpu1;
 	struct device_node *np_codec;
 	struct snd_soc_dai_link *dai;
 	int i, index;
@@ -326,7 +379,8 @@ static int rockchip_sound_of_parse_dais(struct device *dev,
 	if (!card->dai_link)
 		return -ENOMEM;
 
-	np_cpu = of_parse_phandle(dev->of_node, "rockchip,cpu", 0);
+	np_cpu0 = of_parse_phandle(dev->of_node, "rockchip,cpu", 0);
+	np_cpu1 = of_parse_phandle(dev->of_node, "rockchip,cpu", 1);
 
 	card->num_links = 0;
 	for (i = 0; i < ARRAY_SIZE(rockchip_dais); i++) {
@@ -342,6 +396,7 @@ static int rockchip_sound_of_parse_dais(struct device *dev,
 		if (index < 0)
 			continue;
 
+		np_cpu = (index == DAILINK_CDNDP) ? np_cpu1 : np_cpu0;
 		if (!np_cpu) {
 			dev_err(dev, "Missing 'rockchip,cpu' for %s\n",
 				rockchip_dais[index].name);
-- 
cgit v1.2.3


From 626d84db64d746f63eadf5e241fc2211419e11b8 Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Thu, 24 Aug 2017 12:52:26 +0800
Subject: ASoC: rockchip: Add support for DMIC codec

Add support for optional dmic codec.

Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/Kconfig            |  1 +
 sound/soc/rockchip/rk3399_gru_sound.c | 36 +++++++++++++++++++++++++++++++++++
 2 files changed, 37 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
index 8f0d0d8d34e6..b0825370d262 100644
--- a/sound/soc/rockchip/Kconfig
+++ b/sound/soc/rockchip/Kconfig
@@ -69,6 +69,7 @@ config SND_SOC_RK3399_GRU_SOUND
 	select SND_SOC_DA7219
 	select SND_SOC_RT5514_SPI
 	select SND_SOC_HDMI_CODEC
+	select SND_SOC_DMIC
 	help
 	  Say Y or M here if you want to add support multiple codecs for SoC
 	  audio on Rockchip RK3399 GRU boards.
diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index 91aab5c18f3b..5ab25962cabd 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -267,6 +267,28 @@ static int rockchip_sound_cdndp_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	unsigned int mclk;
+	int ret;
+
+	mclk = params_rate(params) * SOUND_FS;
+
+	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0);
+	if (ret) {
+		dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n",
+				__func__, mclk, ret);
+		return ret;
+	}
+
+	/* Wait for DMIC stable */
+	msleep(dmic_wakeup_delay);
+
+	return 0;
+}
+
 static const struct snd_soc_ops rockchip_sound_max98357a_ops = {
 	.hw_params = rockchip_sound_max98357a_hw_params,
 };
@@ -283,6 +305,10 @@ static struct snd_soc_ops rockchip_sound_cdndp_ops = {
 	.hw_params = rockchip_sound_cdndp_hw_params,
 };
 
+static struct snd_soc_ops rockchip_sound_dmic_ops = {
+	.hw_params = rockchip_sound_dmic_hw_params,
+};
+
 static struct snd_soc_card rockchip_sound_card = {
 	.name = "rk3399-gru-sound",
 	.owner = THIS_MODULE,
@@ -297,6 +323,7 @@ static struct snd_soc_card rockchip_sound_card = {
 enum {
 	DAILINK_CDNDP,
 	DAILINK_DA7219,
+	DAILINK_DMIC,
 	DAILINK_MAX98357A,
 	DAILINK_RT5514,
 	DAILINK_RT5514_DSP,
@@ -305,6 +332,7 @@ enum {
 static const char * const dailink_compat[] = {
 	[DAILINK_CDNDP] = "rockchip,rk3399-cdn-dp",
 	[DAILINK_DA7219] = "dlg,da7219",
+	[DAILINK_DMIC] = "dmic-codec",
 	[DAILINK_MAX98357A] = "maxim,max98357a",
 	[DAILINK_RT5514] = "realtek,rt5514-i2c",
 	[DAILINK_RT5514_DSP] = "realtek,rt5514-spi",
@@ -329,6 +357,14 @@ static const struct snd_soc_dai_link rockchip_dais[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
 	},
+	[DAILINK_DMIC] = {
+		.name = "DMIC",
+		.stream_name = "DMIC PCM",
+		.codec_dai_name = "dmic-hifi",
+		.ops = &rockchip_sound_dmic_ops,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+	},
 	[DAILINK_MAX98357A] = {
 		.name = "MAX98357A",
 		.stream_name = "MAX98357A PCM",
-- 
cgit v1.2.3


From 1bb7cb68feb73a8106546cbd3379590297b0a950 Mon Sep 17 00:00:00 2001
From: Oskar Schirmer
Date: Fri, 25 Aug 2017 21:36:56 +0000
Subject: ASoC: tas2552: Fix fraction overflow in PLL calculation

Setting the PLL involves the calculation of a fixed point ratio
with 4 decimal digits fraction, referred to as "J.D". The
fraction "D" is stored separately from the integer part "J"
and is limited to 0..9999.

The current algorithm uses integer registers to calculate the
fraction part, but failed to compensate for rounding errors,
resulting in values larger than 9999 for the fraction part
occasionally, e.g. for 44.1kHz audio rate and pll_clkin =
3763400 it would set J to 11 and D to 10002, which will at
best result in wrong pitch.

The critical part is the "pll_clkin / 10000", which would be
ok with real numbers, but using integer arithmetic the rounding
decreases the divisor, thus increasing the final quotient.

The issue is solved by linear interpolation over the reciprocal
function between the two adjacent points with integer divisor,
i.e. pll_clkin / 10000 and pll_clkin / 10000 + 1, and doing
all rounding to the lower result.

As a side effect to the bug fix, the approximation to the
desired frequency is much better, for the above mentioned
example we get 11.9993, while the true ratio is 11.9993623.

Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2552.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 8840f72f3c4a..10116169a239 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -192,7 +192,7 @@ static int tas2552_setup_pll(struct snd_soc_codec *codec,
 		 * pll_clk = (.5 * pll_clkin * J.D) / 2^p
 		 * Need to fill in J and D here based on incoming freq
 		 */
-		unsigned int d;
+		unsigned int d, q, t;
 		u8 j;
 		u8 pll_sel = (tas2552->pll_clk_id << 3) & TAS2552_PLL_SRC_MASK;
 		u8 p = snd_soc_read(codec, TAS2552_PLL_CTRL_1);
@@ -200,9 +200,12 @@ static int tas2552_setup_pll(struct snd_soc_codec *codec,
 		p = (p >> 7);
 
 recalc:
-		j = (pll_clk * 2 * (1 << p)) / pll_clkin;
-		d = (pll_clk * 2 * (1 << p)) % pll_clkin;
-		d /= (pll_clkin / 10000);
+		t = (pll_clk * 2) << p;
+		j = t / pll_clkin;
+		d = t % pll_clkin;
+		t = pll_clkin / 10000;
+		q = d / (t + 1);
+		d = q + ((9999 - pll_clkin % 10000) * (d / t - q)) / 10000;
 
 		if (d && (pll_clkin < 512000 || pll_clkin > 9200000)) {
 			if (tas2552->pll_clk_id == TAS2552_PLL_CLKIN_BCLK) {
-- 
cgit v1.2.3


From 60b0fdeac19945107b956a8a2a0ba3a14827db60 Mon Sep 17 00:00:00 2001
From: Bard Liao
Date: Wed, 30 Aug 2017 12:28:31 +0800
Subject: ASoC: rt274: add acpi id

Add ACPI id for Intel platform.

Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt274.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index fb683ffc4e03..0be69c112c45 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -1102,6 +1102,7 @@ MODULE_DEVICE_TABLE(i2c, rt274_i2c_id);
 
 static const struct acpi_device_id rt274_acpi_match[] = {
 	{ "10EC0274", 0 },
+	{ "INT34C2", 0 },
 	{},
 };
 MODULE_DEVICE_TABLE(acpi, rt274_acpi_match);
-- 
cgit v1.2.3


From 9c324afe3b90e406b15938acafaa2636a7e85ff3 Mon Sep 17 00:00:00 2001
From: oder_chiou@realtek.com
Date: Wed, 30 Aug 2017 13:08:31 +0800
Subject: ASoC: rt5663: Add delay for jack plug in

Codec initialize takes some time when 3.5mm jack plugged in. Add a
delay to report jack plugged event to user space to avoid pop noise.

Signed-off-by: Hsinyu Chao <hychao@chromium.org>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5663.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index fdc2bbcb67c2..14716d25173b 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -1705,6 +1705,9 @@ static void rt5663_jack_detect_work(struct work_struct *work)
 			default:
 				dev_err(codec->dev, "Unknown CODEC Version\n");
 			}
+
+			/* Delay the jack insert report to avoid pop noise */
+			msleep(30);
 		} else {
 			/* jack is already in, report button event */
 			report = SND_JACK_HEADSET;
-- 
cgit v1.2.3


From 1fe165b7e9c3e1b1cfa98310979db0814c7445ea Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Wed, 30 Aug 2017 17:58:14 +0800
Subject: ASoC: rockchip: constify snd_soc_ops structures

Add const to dp/dmic snd_soc_ops.

Fixes: 626d84db64d7 (ASoC: rockchip: Add support for DMIC codec)
Fixes: 3313faf1053e (ASoC: rockchip: Add support for DP codec)
Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rk3399_gru_sound.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index 5ab25962cabd..0513fe480353 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -301,11 +301,11 @@ static const struct snd_soc_ops rockchip_sound_da7219_ops = {
 	.hw_params = rockchip_sound_da7219_hw_params,
 };
 
-static struct snd_soc_ops rockchip_sound_cdndp_ops = {
+static const struct snd_soc_ops rockchip_sound_cdndp_ops = {
 	.hw_params = rockchip_sound_cdndp_hw_params,
 };
 
-static struct snd_soc_ops rockchip_sound_dmic_ops = {
+static const struct snd_soc_ops rockchip_sound_dmic_ops = {
 	.hw_params = rockchip_sound_dmic_hw_params,
 };
 
-- 
cgit v1.2.3


From 943fa0228252320de503b923dfe46334f755aaae Mon Sep 17 00:00:00 2001
From: Jeffy Chen
Date: Sat, 12 Aug 2017 08:40:38 +0800
Subject: ASoC: hdmi-codec: Use different name for playback streams

Currently the hdmi i2s playback stream and hdmi spdif playback stream
are using the same name. So when they are enabled at the same time,
kernel will print this warning:

[    2.201835] hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: Failed to
create Playback debugfs file

Assign different names to them to avoid that.

Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdmi-codec.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index f7520413c671..189af068f28d 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -696,7 +696,7 @@ static struct snd_soc_dai_driver hdmi_i2s_dai = {
 	.name = "i2s-hifi",
 	.id = DAI_ID_I2S,
 	.playback = {
-		.stream_name = "Playback",
+		.stream_name = "I2S Playback",
 		.channels_min = 2,
 		.channels_max = 8,
 		.rates = HDMI_RATES,
@@ -711,7 +711,7 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = {
 	.name = "spdif-hifi",
 	.id = DAI_ID_SPDIF,
 	.playback = {
-		.stream_name = "Playback",
+		.stream_name = "SPDIF Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = HDMI_RATES,
-- 
cgit v1.2.3


From 71ccef0df533cd9c8c6cbf1483a636a092088ab9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 24 Aug 2017 00:57:35 +0000
Subject: ASoC: add Component level set_sysclk

In current ALSA SoC, Codec only has set_sysclk feature.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 11 +++++++++++
 sound/soc/soc-core.c | 45 ++++++++++++++++++++++++++++++++++++++++-----
 2 files changed, 51 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index fbf57ff75958..813cdb95e049 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -795,6 +795,10 @@ struct snd_soc_component_driver {
 	int (*suspend)(struct snd_soc_component *);
 	int (*resume)(struct snd_soc_component *);
 
+	/* component wide operations */
+	int (*set_sysclk)(struct snd_soc_component *component,
+			  int clk_id, int source, unsigned int freq, int dir);
+
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
 				 struct of_phandle_args *args,
@@ -865,6 +869,9 @@ struct snd_soc_component {
 	int (*suspend)(struct snd_soc_component *);
 	int (*resume)(struct snd_soc_component *);
 
+	int (*set_sysclk)(struct snd_soc_component *component,
+			  int clk_id, int source, unsigned int freq, int dir);
+
 	/* machine specific init */
 	int (*init)(struct snd_soc_component *component);
 
@@ -1459,6 +1466,10 @@ void snd_soc_component_async_complete(struct snd_soc_component *component);
 int snd_soc_component_test_bits(struct snd_soc_component *component,
 	unsigned int reg, unsigned int mask, unsigned int value);
 
+/* component wide operations */
+int snd_soc_component_set_sysclk(struct snd_soc_component *component,
+			int clk_id, int source, unsigned int freq, int dir);
+
 #ifdef CONFIG_REGMAP
 
 void snd_soc_component_init_regmap(struct snd_soc_component *component,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 10efd15338fb..654db550039d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2590,11 +2590,9 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 {
 	if (dai->driver && dai->driver->ops->set_sysclk)
 		return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
-	else if (dai->codec && dai->codec->driver->set_sysclk)
-		return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0,
-						      freq, dir);
-	else
-		return -ENOTSUPP;
+
+	return snd_soc_component_set_sysclk(dai->component, clk_id, 0,
+					    freq, dir);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 
@@ -2619,6 +2617,32 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk);
 
+/**
+ * snd_soc_component_set_sysclk - configure COMPONENT system or master clock.
+ * @component: COMPONENT
+ * @clk_id: DAI specific clock ID
+ * @source: Source for the clock
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id,
+			     int source, unsigned int freq, int dir)
+{
+	/* will be removed */
+	if (component->set_sysclk)
+		return component->set_sysclk(component, clk_id, source,
+					     freq, dir);
+
+	if (component->driver->set_sysclk)
+		return component->driver->set_sysclk(component, clk_id, source,
+						 freq, dir);
+
+	return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_set_sysclk);
+
 /**
  * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
  * @dai: DAI
@@ -3174,6 +3198,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	component->remove = component->driver->remove;
 	component->suspend = component->driver->suspend;
 	component->resume = component->driver->resume;
+	component->set_sysclk = component->driver->set_sysclk;
 
 	dapm = &component->dapm;
 	dapm->dev = dev;
@@ -3557,6 +3582,14 @@ static int snd_soc_codec_drv_read(struct snd_soc_component *component,
 	return 0;
 }
 
+static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component,
+			  int clk_id, int source, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
+
+	return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir);
+}
+
 static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm,
 	enum snd_soc_bias_level level)
 {
@@ -3608,6 +3641,8 @@ int snd_soc_register_codec(struct device *dev,
 		codec->component.write = snd_soc_codec_drv_write;
 	if (codec_drv->read)
 		codec->component.read = snd_soc_codec_drv_read;
+	if (codec_drv->set_sysclk)
+		codec->component.set_sysclk = snd_soc_codec_set_sysclk_;
 	codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
 
 	dapm = snd_soc_codec_get_dapm(codec);
-- 
cgit v1.2.3


From ef641e5d5e6c7a07748239036b786a90ba9b9a95 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 24 Aug 2017 00:57:51 +0000
Subject: ASoC: add Component level set_pll

In current ALSA SoC, Codec only has set_pll feature.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  7 +++++++
 sound/soc/soc-core.c | 47 ++++++++++++++++++++++++++++++++++++++++++-----
 2 files changed, 49 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 813cdb95e049..23681538c849 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -798,6 +798,8 @@ struct snd_soc_component_driver {
 	/* component wide operations */
 	int (*set_sysclk)(struct snd_soc_component *component,
 			  int clk_id, int source, unsigned int freq, int dir);
+	int (*set_pll)(struct snd_soc_component *component, int pll_id,
+		       int source, unsigned int freq_in, unsigned int freq_out);
 
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
@@ -871,6 +873,8 @@ struct snd_soc_component {
 
 	int (*set_sysclk)(struct snd_soc_component *component,
 			  int clk_id, int source, unsigned int freq, int dir);
+	int (*set_pll)(struct snd_soc_component *component, int pll_id,
+		       int source, unsigned int freq_in, unsigned int freq_out);
 
 	/* machine specific init */
 	int (*init)(struct snd_soc_component *component);
@@ -1469,6 +1473,9 @@ int snd_soc_component_test_bits(struct snd_soc_component *component,
 /* component wide operations */
 int snd_soc_component_set_sysclk(struct snd_soc_component *component,
 			int clk_id, int source, unsigned int freq, int dir);
+int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
+			      int source, unsigned int freq_in,
+			      unsigned int freq_out);
 
 #ifdef CONFIG_REGMAP
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 654db550039d..ee641cd8aa22 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2679,11 +2679,9 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
 	if (dai->driver && dai->driver->ops->set_pll)
 		return dai->driver->ops->set_pll(dai, pll_id, source,
 					 freq_in, freq_out);
-	else if (dai->codec && dai->codec->driver->set_pll)
-		return dai->codec->driver->set_pll(dai->codec, pll_id, source,
-						   freq_in, freq_out);
-	else
-		return -EINVAL;
+
+	return snd_soc_component_set_pll(dai->component, pll_id, source,
+					 freq_in, freq_out);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
 
@@ -2708,6 +2706,33 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll);
 
+/*
+ * snd_soc_component_set_pll - configure component PLL.
+ * @component: COMPONENT
+ * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
+			      int source, unsigned int freq_in,
+			      unsigned int freq_out)
+{
+	/* will be removed */
+	if (component->set_pll)
+		return component->set_pll(component, pll_id, source,
+					      freq_in, freq_out);
+
+	if (component->driver->set_pll)
+		return component->driver->set_pll(component, pll_id, source,
+					      freq_in, freq_out);
+
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_set_pll);
+
 /**
  * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio.
  * @dai: DAI
@@ -3199,6 +3224,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	component->suspend = component->driver->suspend;
 	component->resume = component->driver->resume;
 	component->set_sysclk = component->driver->set_sysclk;
+	component->set_pll = component->driver->set_pll;
 
 	dapm = &component->dapm;
 	dapm->dev = dev;
@@ -3590,6 +3616,15 @@ static int snd_soc_codec_set_sysclk_(struct snd_soc_component *component,
 	return snd_soc_codec_set_sysclk(codec, clk_id, source, freq, dir);
 }
 
+static int snd_soc_codec_set_pll_(struct snd_soc_component *component,
+				  int pll_id, int source, unsigned int freq_in,
+				  unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
+
+	return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out);
+}
+
 static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm,
 	enum snd_soc_bias_level level)
 {
@@ -3643,6 +3678,8 @@ int snd_soc_register_codec(struct device *dev,
 		codec->component.read = snd_soc_codec_drv_read;
 	if (codec_drv->set_sysclk)
 		codec->component.set_sysclk = snd_soc_codec_set_sysclk_;
+	if (codec_drv->set_pll)
+		codec->component.set_pll = snd_soc_codec_set_pll_;
 	codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
 
 	dapm = snd_soc_codec_get_dapm(codec);
-- 
cgit v1.2.3


From 44c07365e9e2c87c7e04d63293618c391d1480ec Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Thu, 24 Aug 2017 00:58:07 +0000
Subject: ASoC: add Component level set_jack

In current ALSA SoC, Codec only has set_jack feature.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  4 ++++
 sound/soc/soc-core.c | 11 +++++++++++
 sound/soc/soc-jack.c | 22 ++++++++++++++++++++++
 3 files changed, 37 insertions(+)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 23681538c849..feb896815069 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -800,6 +800,8 @@ struct snd_soc_component_driver {
 			  int clk_id, int source, unsigned int freq, int dir);
 	int (*set_pll)(struct snd_soc_component *component, int pll_id,
 		       int source, unsigned int freq_in, unsigned int freq_out);
+	int (*set_jack)(struct snd_soc_component *component,
+			struct snd_soc_jack *jack,  void *data);
 
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
@@ -875,6 +877,8 @@ struct snd_soc_component {
 			  int clk_id, int source, unsigned int freq, int dir);
 	int (*set_pll)(struct snd_soc_component *component, int pll_id,
 		       int source, unsigned int freq_in, unsigned int freq_out);
+	int (*set_jack)(struct snd_soc_component *component,
+			struct snd_soc_jack *jack,  void *data);
 
 	/* machine specific init */
 	int (*init)(struct snd_soc_component *component);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ee641cd8aa22..3d7287858609 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3225,6 +3225,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	component->resume = component->driver->resume;
 	component->set_sysclk = component->driver->set_sysclk;
 	component->set_pll = component->driver->set_pll;
+	component->set_jack = component->driver->set_jack;
 
 	dapm = &component->dapm;
 	dapm->dev = dev;
@@ -3625,6 +3626,14 @@ static int snd_soc_codec_set_pll_(struct snd_soc_component *component,
 	return snd_soc_codec_set_pll(codec, pll_id, source, freq_in, freq_out);
 }
 
+static int snd_soc_codec_set_jack_(struct snd_soc_component *component,
+			       struct snd_soc_jack *jack, void *data)
+{
+	struct snd_soc_codec *codec = snd_soc_component_to_codec(component);
+
+	return snd_soc_codec_set_jack(codec, jack, data);
+}
+
 static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm,
 	enum snd_soc_bias_level level)
 {
@@ -3680,6 +3689,8 @@ int snd_soc_register_codec(struct device *dev,
 		codec->component.set_sysclk = snd_soc_codec_set_sysclk_;
 	if (codec_drv->set_pll)
 		codec->component.set_pll = snd_soc_codec_set_pll_;
+	if (codec_drv->set_jack)
+		codec->component.set_jack = snd_soc_codec_set_jack_;
 	codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
 
 	dapm = snd_soc_codec_get_dapm(codec);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7daf21fee355..2f9f496ab14f 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -40,6 +40,28 @@ int snd_soc_codec_set_jack(struct snd_soc_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack);
 
+/**
+ * snd_soc_component_set_jack - configure component jack.
+ * @component: COMPONENTs
+ * @jack: structure to use for the jack
+ * @data: can be used if codec driver need extra data for configuring jack
+ *
+ * Configures and enables jack detection function.
+ */
+int snd_soc_component_set_jack(struct snd_soc_component *component,
+			       struct snd_soc_jack *jack, void *data)
+{
+	/* will be removed */
+	if (component->set_jack)
+		return component->set_jack(component, jack, data);
+
+	if (component->driver->set_jack)
+		return component->driver->set_jack(component, jack, data);
+
+	return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_set_jack);
+
 /**
  * snd_soc_card_jack_new - Create a new jack
  * @card:  ASoC card
-- 
cgit v1.2.3


From 19b0fa11bb1c0c24296c9f670a37c091bd3c815d Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi
Date: Thu, 31 Aug 2017 11:49:47 +0300
Subject: ASoC: tlv320aic3x: Support for OCMV configuration

In aic3x class of devices Output Common-Mode Voltage can be configured for
better analog performance.
The OCMV value depends on the Analog and digital domain power supply
voltage configuration.

The default OCMV of 1.35V gives best performance when AVDD is around 2.7V
and DVDD is 1.525V, but for higher AVDD/DVDD higher OCMV setting is
recommended.

The patch gives an automatic way of guessing the best OCMV which can be
overwritten by a DT parameter if needed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/tlv320aic3x.txt      |  5 +++
 sound/soc/codecs/tlv320aic3x.c                     | 45 ++++++++++++++++++++++
 sound/soc/codecs/tlv320aic3x.h                     |  8 ++++
 3 files changed, 58 insertions(+)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index 47a213c411ce..ba5b45c483f5 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -26,6 +26,11 @@ Optional properties:
 	3 - MICBIAS output is connected to AVDD,
 	If this node is not mentioned or if the value is incorrect, then MicBias
 	is powered down.
+- ai3x-ocmv - Output Common-Mode Voltage selection:
+	0 - 1.35V,
+	1 - 1.5V,
+	2 - 1.65V,
+	3 - 1.8V
 - AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
   device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 29bf8c81ae02..ca880859f0c3 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -93,6 +93,8 @@ struct aic3x_priv {
 
 	/* Selects the micbias voltage */
 	enum aic3x_micbias_voltage micbias_vg;
+	/* Output Common-Mode Voltage */
+	u8 ocmv;
 };
 
 static const struct reg_default aic3x_reg[] = {
@@ -1572,6 +1574,10 @@ static int aic3x_init(struct snd_soc_codec *codec)
 		break;
 	}
 
+	/*  Output common-mode voltage = 1.5 V */
+	snd_soc_update_bits(codec, HPOUT_SC, HPOUT_SC_OCMV_MASK,
+			    aic3x->ocmv << HPOUT_SC_OCMV_SHIFT);
+
 	return 0;
 }
 
@@ -1699,6 +1705,43 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
 	},
 };
 
+static void aic3x_configure_ocmv(struct i2c_client *client)
+{
+	struct device_node *np = client->dev.of_node;
+	struct aic3x_priv *aic3x = i2c_get_clientdata(client);
+	u32 value;
+	int dvdd, avdd;
+
+	if (np && !of_property_read_u32(np, "ai3x-ocmv", &value)) {
+		/* OCMV setting is forced by DT */
+		if (value <= 3) {
+			aic3x->ocmv = value;
+			return;
+		}
+	}
+
+	dvdd = regulator_get_voltage(aic3x->supplies[1].consumer);
+	avdd = regulator_get_voltage(aic3x->supplies[2].consumer);
+
+	if (avdd > 3600000 || dvdd > 1950000) {
+		dev_warn(&client->dev,
+			 "Too high supply voltage(s) AVDD: %d, DVDD: %d\n",
+			 avdd, dvdd);
+	} else if (avdd == 3600000 && dvdd == 1950000) {
+		aic3x->ocmv = HPOUT_SC_OCMV_1_8V;
+	} else if (avdd > 3300000 && dvdd > 1800000) {
+		aic3x->ocmv = HPOUT_SC_OCMV_1_65V;
+	} else if (avdd > 3000000 && dvdd > 1650000) {
+		aic3x->ocmv = HPOUT_SC_OCMV_1_5V;
+	} else if (avdd >= 2700000 && dvdd >= 1525000) {
+		aic3x->ocmv = HPOUT_SC_OCMV_1_35V;
+	} else {
+		dev_warn(&client->dev,
+			 "Invalid supply voltage(s) AVDD: %d, DVDD: %d\n",
+			 avdd, dvdd);
+	}
+}
+
 /*
  * AIC3X 2 wire address can be up to 4 devices with device addresses
  * 0x18, 0x19, 0x1A, 0x1B
@@ -1816,6 +1859,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 		goto err_gpio;
 	}
 
+	aic3x_configure_ocmv(i2c);
+
 	if (aic3x->model == AIC3X_MODEL_3007) {
 		ret = regmap_register_patch(aic3x->regmap, aic3007_class_d,
 					    ARRAY_SIZE(aic3007_class_d));
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 89fa692df206..34c35196aa0d 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -243,6 +243,14 @@
 #define MICBIAS_LEVEL_SHIFT	(6)
 #define MICBIAS_LEVEL_MASK	(3 << 6)
 
+/* HPOUT_SC */
+#define HPOUT_SC_OCMV_MASK	(3 << 6)
+#define HPOUT_SC_OCMV_SHIFT	(6)
+#define HPOUT_SC_OCMV_1_35V	0
+#define HPOUT_SC_OCMV_1_5V	1
+#define HPOUT_SC_OCMV_1_65V	2
+#define HPOUT_SC_OCMV_1_8V	3
+
 /* headset detection / button API */
 
 /* The AIC3x supports detection of stereo headsets (GND + left + right signal)
-- 
cgit v1.2.3


From 88faae2c2e35c88a806c052a7fc20948211b0895 Mon Sep 17 00:00:00 2001
From: Ian W MORRISON
Date: Thu, 31 Aug 2017 08:35:47 +1000
Subject: ASoC: rt5645: Add jack detection workaround for MINIX Z83-4 based
 devices

The MINIX NEO Z83-4 and MINIX NEO Z83-4 Pro devices requires jd_mode=3
to make the jack detection work. Using a BIOS DMI product of "Z83-4"
will match both devices of 'NEO Z83-4' and 'Z83-4 Pro'.

Signed-off-by: Ian W Morrison <ianwmorrison@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 17 +++++++++++++++++
 1 file changed, 17 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index d2f038001908..2e145f089d9d 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3655,6 +3655,21 @@ static struct dmi_system_id dmi_platform_asus_t100ha[] = {
 	{ }
 };
 
+static struct rt5645_platform_data minix_z83_4_platform_data = {
+	.jd_mode = 3,
+};
+
+static struct dmi_system_id dmi_platform_minix_z83_4[] = {
+	{
+		.ident = "MINIX Z83-4",
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Z83-4"),
+		},
+	},
+	{ }
+};
+
 static bool rt5645_check_dp(struct device *dev)
 {
 	if (device_property_present(dev, "realtek,in2-differential") ||
@@ -3709,6 +3724,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 		rt5645->pdata = gpd_win_platform_data;
 	else if (dmi_check_system(dmi_platform_asus_t100ha))
 		rt5645->pdata = general_platform_data2;
+	else if (dmi_check_system(dmi_platform_minix_z83_4))
+		rt5645->pdata = minix_z83_4_platform_data;
 
 	if (quirk != -1) {
 		rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk);
-- 
cgit v1.2.3


From 0515760fa1159ffa863c7b2b73466aaff7d11a80 Mon Sep 17 00:00:00 2001
From: Christophe Jaillet
Date: Thu, 31 Aug 2017 06:40:42 +0200
Subject: ALSA: ac97c: Fix an error handling path in 'atmel_ac97c_probe()'

If 'clk_prepare_enable()' fails, we must release some resources before
returning. Add a new label in the existing error handling path and 'goto'
there.

Fixes: 260ea95cc027 ("ASoC: atmel: ac97c: Handle return value of clk_prepare_enable.")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/atmel/ac97c.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 52b0522fda20..d78405329ceb 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -785,7 +785,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev)
 	}
 	retval = clk_prepare_enable(pclk);
 	if (retval)
-		return retval;
+		goto err_prepare_enable;
 
 	retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1,
 			      SNDRV_DEFAULT_STR1, THIS_MODULE,
@@ -881,6 +881,7 @@ err_request_irq:
 	snd_card_free(card);
 err_snd_card_new:
 	clk_disable_unprepare(pclk);
+err_prepare_enable:
 	clk_put(pclk);
 	return retval;
 }
-- 
cgit v1.2.3


From 9c1743ebe0da614517a1b94f171924d5fd758df4 Mon Sep 17 00:00:00 2001
From: Ryan Lee
Date: Mon, 28 Aug 2017 16:30:55 -0700
Subject: ASoC: max98927: Updated volatile register list

Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98927.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index b5ee29499e16..768e84dbcdf6 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -495,6 +495,13 @@ static bool max98927_volatile_reg(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3:
+	case MAX98927_R004C_MEAS_ADC_CH0_READ:
+	case MAX98927_R004D_MEAS_ADC_CH1_READ:
+	case MAX98927_R004E_MEAS_ADC_CH2_READ:
+	case MAX98927_R0051_BROWNOUT_STATUS:
+	case MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ:
+	case MAX98927_R01FF_REV_ID:
+	case MAX98927_R0100_SOFT_RESET:
 		return true;
 	default:
 		return false;
-- 
cgit v1.2.3


From 01c6f1b5c832b75af5d34555201050db896285a8 Mon Sep 17 00:00:00 2001
From: Ryan Lee
Date: Mon, 28 Aug 2017 16:30:56 -0700
Subject: ASoC: max98927: Added missing \n to end of dev_err messages

Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98927.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index 768e84dbcdf6..f74cdb065bf2 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -159,7 +159,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		mode = MAX98927_PCM_MASTER_MODE_MASTER;
 		break;
 	default:
-		dev_err(codec->dev, "DAI clock mode unsupported");
+		dev_err(codec->dev, "DAI clock mode unsupported\n");
 		return -EINVAL;
 	}
 
@@ -175,7 +175,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		invert = MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE;
 		break;
 	default:
-		dev_err(codec->dev, "DAI invert mode unsupported");
+		dev_err(codec->dev, "DAI invert mode unsupported\n");
 		return -EINVAL;
 	}
 
@@ -311,7 +311,7 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream,
 		chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32;
 		break;
 	default:
-		dev_err(codec->dev, "format unsupported %d",
+		dev_err(codec->dev, "format unsupported %d\n",
 			params_format(params));
 		goto err;
 	}
-- 
cgit v1.2.3


From 848844b87a26265019215dbb7fb588a5fac0a110 Mon Sep 17 00:00:00 2001
From: Ryan Lee
Date: Mon, 28 Aug 2017 16:30:58 -0700
Subject: ASoC: max98927: Modified chip default register values

Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98927.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index f74cdb065bf2..5b853604120a 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -44,9 +44,9 @@ static struct reg_default max98927_reg[] = {
 	{MAX98927_R0011_CLK_MON,  0x00},
 	{MAX98927_R0012_WDOG_CTRL,  0x00},
 	{MAX98927_R0013_WDOG_RST,  0x00},
-	{MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH,  0x00},
-	{MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH,  0x00},
-	{MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS,  0x00},
+	{MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH,  0x75},
+	{MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH,  0x8c},
+	{MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS,  0x08},
 	{MAX98927_R0017_PIN_CFG,  0x55},
 	{MAX98927_R0018_PCM_RX_EN_A,  0x00},
 	{MAX98927_R0019_PCM_RX_EN_B,  0x00},
@@ -82,14 +82,14 @@ static struct reg_default max98927_reg[] = {
 	{MAX98927_R003A_AMP_EN,  0x00},
 	{MAX98927_R003B_SPK_SRC_SEL,  0x00},
 	{MAX98927_R003C_SPK_GAIN,  0x00},
-	{MAX98927_R003D_SSM_CFG,  0x01},
+	{MAX98927_R003D_SSM_CFG,  0x04},
 	{MAX98927_R003E_MEAS_EN,  0x00},
 	{MAX98927_R003F_MEAS_DSP_CFG,  0x04},
 	{MAX98927_R0040_BOOST_CTRL0,  0x00},
 	{MAX98927_R0041_BOOST_CTRL3,  0x00},
 	{MAX98927_R0042_BOOST_CTRL1,  0x00},
 	{MAX98927_R0043_MEAS_ADC_CFG,  0x00},
-	{MAX98927_R0044_MEAS_ADC_BASE_MSB,  0x00},
+	{MAX98927_R0044_MEAS_ADC_BASE_MSB,  0x01},
 	{MAX98927_R0045_MEAS_ADC_BASE_LSB,  0x00},
 	{MAX98927_R0046_ADC_CH0_DIVIDE,  0x00},
 	{MAX98927_R0047_ADC_CH1_DIVIDE,  0x00},
-- 
cgit v1.2.3


From f81991ddd001981bbe4ea0d25f88a4f95fb018be Mon Sep 17 00:00:00 2001
From: Ryan Lee
Date: Mon, 28 Aug 2017 16:30:59 -0700
Subject: ASoC: max98927: Added PM suspend and resume function

Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98927.c | 27 ++++++++++++++++++++++++++-
 1 file changed, 26 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index 5b853604120a..caf5340ca6bb 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -701,6 +701,31 @@ static int max98927_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int max98927_suspend(struct device *dev)
+{
+	struct max98927_priv *max98927 = dev_get_drvdata(dev);
+
+	regcache_cache_only(max98927->regmap, true);
+	regcache_mark_dirty(max98927->regmap);
+	return 0;
+}
+static int max98927_resume(struct device *dev)
+{
+	struct max98927_priv *max98927 = dev_get_drvdata(dev);
+
+	regmap_write(max98927->regmap,
+		MAX98927_R0100_SOFT_RESET, MAX98927_SOFT_RESET);
+	regcache_cache_only(max98927->regmap, false);
+	regcache_sync(max98927->regmap);
+	return 0;
+}
+#endif
+
+static const struct dev_pm_ops max98927_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(max98927_suspend, max98927_resume)
+};
+
 static const struct snd_soc_codec_driver soc_codec_dev_max98927 = {
 	.probe = max98927_probe,
 	.component_driver = {
@@ -834,7 +859,7 @@ static struct i2c_driver max98927_i2c_driver = {
 		.name = "max98927",
 		.of_match_table = of_match_ptr(max98927_of_match),
 		.acpi_match_table = ACPI_PTR(max98927_acpi_match),
-		.pm = NULL,
+		.pm = &max98927_pm,
 	},
 	.probe  = max98927_i2c_probe,
 	.remove = max98927_i2c_remove,
-- 
cgit v1.2.3


From 607f7777bf0c0224bfadd31a37acc8d1fd25324d Mon Sep 17 00:00:00 2001
From: Ryan Lee
Date: Mon, 28 Aug 2017 16:31:00 -0700
Subject: ASoC: max98927: Modified DAPM widget and map to enable/disable VI
 sense path

Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98927.c | 28 +++++++++++++++++-----------
 1 file changed, 17 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index caf5340ca6bb..4bbf2098f255 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -418,11 +418,6 @@ static int max98927_dac_event(struct snd_soc_dapm_widget *w,
 		regmap_update_bits(max98927->regmap,
 			MAX98927_R003A_AMP_EN,
 			MAX98927_AMP_EN_MASK, 1);
-		/* enable VMON and IMON */
-		regmap_update_bits(max98927->regmap,
-			MAX98927_R003E_MEAS_EN,
-			MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN,
-			MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN);
 		regmap_update_bits(max98927->regmap,
 			MAX98927_R00FF_GLOBAL_SHDN,
 			MAX98927_GLOBAL_EN_MASK, 1);
@@ -434,10 +429,6 @@ static int max98927_dac_event(struct snd_soc_dapm_widget *w,
 		regmap_update_bits(max98927->regmap,
 			MAX98927_R003A_AMP_EN,
 			MAX98927_AMP_EN_MASK, 0);
-		/* disable VMON and IMON */
-		regmap_update_bits(max98927->regmap,
-			MAX98927_R003E_MEAS_EN,
-			MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, 0);
 		break;
 	default:
 		return 0;
@@ -456,14 +447,24 @@ static const struct soc_enum dai_sel_enum =
 static const struct snd_kcontrol_new max98927_dai_controls =
 	SOC_DAPM_ENUM("DAI Sel", dai_sel_enum);
 
+static const struct snd_kcontrol_new max98927_vi_control =
+	SOC_DAPM_SINGLE("Switch", MAX98927_R003F_MEAS_DSP_CFG, 2, 1, 0);
+
 static const struct snd_soc_dapm_widget max98927_dapm_widgets[] = {
-	SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", MAX98927_R003A_AMP_EN,
 		0, 0, max98927_dac_event,
 		SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0,
 		&max98927_dai_controls),
 	SND_SOC_DAPM_OUTPUT("BE_OUT"),
+	SND_SOC_DAPM_AIF_OUT("Voltage Sense", "HiFi Capture",  0,
+		MAX98927_R003E_MEAS_EN, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("Current Sense", "HiFi Capture",  0,
+		MAX98927_R003E_MEAS_EN, 1, 0),
+	SND_SOC_DAPM_SWITCH("VI Sense", SND_SOC_NOPM, 0, 0,
+		&max98927_vi_control),
+	SND_SOC_DAPM_SIGGEN("VMON"),
+	SND_SOC_DAPM_SIGGEN("IMON"),
 };
 
 static DECLARE_TLV_DB_SCALE(max98927_spk_tlv, 300, 300, 0);
@@ -550,11 +551,16 @@ static const struct snd_kcontrol_new max98927_snd_controls[] = {
 };
 
 static const struct snd_soc_dapm_route max98927_audio_map[] = {
-	{"Amp Enable", NULL, "DAI_OUT"},
+	/* Plabyack */
 	{"DAI Sel Mux", "Left", "Amp Enable"},
 	{"DAI Sel Mux", "Right", "Amp Enable"},
 	{"DAI Sel Mux", "LeftRight", "Amp Enable"},
 	{"BE_OUT", NULL, "DAI Sel Mux"},
+	/* Capture */
+	{ "VI Sense", "Switch", "VMON" },
+	{ "VI Sense", "Switch", "IMON" },
+	{ "Voltage Sense", NULL, "VI Sense" },
+	{ "Current Sense", NULL, "VI Sense" },
 };
 
 static struct snd_soc_dai_driver max98927_dai[] = {
-- 
cgit v1.2.3


From dea110501745734baf7a90dc1e1b50481ade6191 Mon Sep 17 00:00:00 2001
From: Ryan Lee
Date: Mon, 28 Aug 2017 16:31:01 -0700
Subject: ASoC: max98927: Changed device property read function

Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98927.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index 4bbf2098f255..926871e36132 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -759,14 +759,14 @@ static void max98927_slot_config(struct i2c_client *i2c,
 	struct max98927_priv *max98927)
 {
 	int value;
+	struct device *dev = &i2c->dev;
 
-	if (!of_property_read_u32(i2c->dev.of_node,
-		"vmon-slot-no", &value))
+	if (!device_property_read_u32(dev, "vmon-slot-no", &value))
 		max98927->v_l_slot = value & 0xF;
 	else
 		max98927->v_l_slot = 0;
-	if (!of_property_read_u32(i2c->dev.of_node,
-		"imon-slot-no", &value))
+
+	if (!device_property_read_u32(dev, "imon-slot-no", &value))
 		max98927->i_l_slot = value & 0xF;
 	else
 		max98927->i_l_slot = 1;
-- 
cgit v1.2.3


From a728f56094e7cf60a1dc0642fe86901d1a4dfb4e Mon Sep 17 00:00:00 2001
From: Vitaly Wool
Date: Thu, 17 Aug 2017 13:42:36 +0200
Subject: ASoC: make clock direction configurable in asoc-simple

Some CPU drivers (e. g. davinci-mcasp) may require the system clock to
be configured as OUT, while there's no good way currently to set
SND_SOC_CLK_OUT in simple-soc driver if the clock is fixed-rate.

This patch makes asoc_simple_card_init_dai() initialize clock to
SND_SOCK_CLK_OUT if explicitly stated in the relevant dts file. This
change is transparent and doesn't change the default behavior.

Signed-off-by: Vitaly Wool <vitaly.wool@konsulko.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/simple-card.txt | 3 +++
 include/sound/simple_card_utils.h                       | 1 +
 sound/soc/generic/simple-card-utils.c                   | 9 +++++++--
 3 files changed, 11 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index c7a93931fad2..166f2290233b 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -86,6 +86,9 @@ Optional CPU/CODEC subnodes properties:
 					  in dai startup() and disabled with
 					  clk_disable_unprepare() in dai
 					  shutdown().
+- system-clock-direction-out		: specifies clock direction as 'out' on
+					  initialization. It is useful for some aCPUs with
+					  fixed clocks.
 
 Example 1 - single DAI link:
 
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 42c6a6ac3ce6..7e25afce6566 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -15,6 +15,7 @@
 struct asoc_simple_dai {
 	const char *name;
 	unsigned int sysclk;
+	int clk_direction;
 	int slots;
 	int slot_width;
 	unsigned int tx_slot_mask;
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 26d64fa40c9c..af44c8c52a19 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -196,7 +196,11 @@ int asoc_simple_card_parse_clk(struct device *dev,
 			simple_dai->sysclk = clk_get_rate(clk);
 	}
 
-	dev_dbg(dev, "%s : sysclk = %d\n", name, simple_dai->sysclk);
+	if (of_property_read_bool(node, "system-clock-direction-out"))
+		simple_dai->clk_direction = SND_SOC_CLOCK_OUT;
+
+	dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name,
+		simple_dai->sysclk, simple_dai->clk_direction);
 
 	return 0;
 }
@@ -310,7 +314,8 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai,
 	int ret;
 
 	if (simple_dai->sysclk) {
-		ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk, 0);
+		ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk,
+					     simple_dai->clk_direction);
 		if (ret && ret != -ENOTSUPP) {
 			dev_err(dai->dev, "simple-card: set_sysclk error\n");
 			return ret;
-- 
cgit v1.2.3


From 8f1e5bf9b4408e91942b94c76583eaf098f19382 Mon Sep 17 00:00:00 2001
From: Li Xu
Date: Fri, 18 Aug 2017 11:00:19 -0500
Subject: ASoC: cs43130: Add support for CS43130 codec

Add support for Cirrus Logic CS43130 codec.

Support:
I2S/DSP PCM playback.
DoP/DSD playback.
HP detection and DC/AC impedance measurement.

Signed-off-by: Li Xu <li.xu@cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig   |    6 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/cs43130.c | 2690 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/cs43130.h |  546 +++++++++
 4 files changed, 3244 insertions(+)
 create mode 100644 sound/soc/codecs/cs43130.c
 create mode 100644 sound/soc/codecs/cs43130.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 6c78b0b49b81..ea7a2d540e5e 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS4271_I2C if I2C
 	select SND_SOC_CS4271_SPI if SPI_MASTER
 	select SND_SOC_CS42XX8_I2C if I2C
+	select SND_SOC_CS43130 if I2C
 	select SND_SOC_CS4349 if I2C
 	select SND_SOC_CS47L24 if MFD_CS47L24
 	select SND_SOC_CS53L30 if I2C
@@ -486,6 +487,11 @@ config SND_SOC_CS42XX8_I2C
 	select SND_SOC_CS42XX8
 	select REGMAP_I2C
 
+# Cirrus Logic CS43130 HiFi DAC
+config SND_SOC_CS43130
+        tristate "Cirrus Logic CS43130 CODEC"
+        depends on I2C
+
 # Cirrus Logic CS4349 HiFi DAC
 config SND_SOC_CS4349
 	tristate "Cirrus Logic CS4349 CODEC"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1755a54e3dc9..5030e1f440d4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -53,6 +53,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o
 snd-soc-cs4271-spi-objs := cs4271-spi.o
 snd-soc-cs42xx8-objs := cs42xx8.o
 snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
+snd-soc-cs43130-objs := cs43130.o
 snd-soc-cs4349-objs := cs4349.o
 snd-soc-cs47l24-objs := cs47l24.o
 snd-soc-cs53l30-objs := cs53l30.o
@@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C)	+= snd-soc-cs4271-i2c.o
 obj-$(CONFIG_SND_SOC_CS4271_SPI)	+= snd-soc-cs4271-spi.o
 obj-$(CONFIG_SND_SOC_CS42XX8)	+= snd-soc-cs42xx8.o
 obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o
+obj-$(CONFIG_SND_SOC_CS43130)   += snd-soc-cs43130.o
 obj-$(CONFIG_SND_SOC_CS4349)	+= snd-soc-cs4349.o
 obj-$(CONFIG_SND_SOC_CS47L24)	+= snd-soc-cs47l24.o
 obj-$(CONFIG_SND_SOC_CS53L30)	+= snd-soc-cs53l30.o
diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c
new file mode 100644
index 000000000000..220e30199c5b
--- /dev/null
+++ b/sound/soc/codecs/cs43130.c
@@ -0,0 +1,2690 @@
+/*
+ * cs43130.c  --  CS43130 ALSA Soc Audio driver
+ *
+ * Copyright 2017 Cirrus Logic, Inc.
+ *
+ * Authors: Li Xu <li.xu@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/platform_device.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/of_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.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 <linux/of_gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/pm_runtime.h>
+#include <linux/of_irq.h>
+#include <linux/completion.h>
+#include <linux/mutex.h>
+#include <linux/workqueue.h>
+#include <sound/jack.h>
+
+#include "cs43130.h"
+
+static const struct reg_default cs43130_reg_defaults[] = {
+	{CS43130_SYS_CLK_CTL_1, 0x06},
+	{CS43130_SP_SRATE, 0x01},
+	{CS43130_SP_BITSIZE, 0x05},
+	{CS43130_PAD_INT_CFG, 0x03},
+	{CS43130_PWDN_CTL, 0xFE},
+	{CS43130_CRYSTAL_SET, 0x04},
+	{CS43130_PLL_SET_1, 0x00},
+	{CS43130_PLL_SET_2, 0x00},
+	{CS43130_PLL_SET_3, 0x00},
+	{CS43130_PLL_SET_4, 0x00},
+	{CS43130_PLL_SET_5, 0x40},
+	{CS43130_PLL_SET_6, 0x10},
+	{CS43130_PLL_SET_7, 0x80},
+	{CS43130_PLL_SET_8, 0x03},
+	{CS43130_PLL_SET_9, 0x02},
+	{CS43130_PLL_SET_10, 0x02},
+	{CS43130_CLKOUT_CTL, 0x00},
+	{CS43130_ASP_NUM_1, 0x01},
+	{CS43130_ASP_NUM_2, 0x00},
+	{CS43130_ASP_DEN_1, 0x08},
+	{CS43130_ASP_DEN_2, 0x00},
+	{CS43130_ASP_LRCK_HI_TIME_1, 0x1F},
+	{CS43130_ASP_LRCK_HI_TIME_2, 0x00},
+	{CS43130_ASP_LRCK_PERIOD_1, 0x3F},
+	{CS43130_ASP_LRCK_PERIOD_2, 0x00},
+	{CS43130_ASP_CLOCK_CONF, 0x0C},
+	{CS43130_ASP_FRAME_CONF, 0x0A},
+	{CS43130_XSP_NUM_1, 0x01},
+	{CS43130_XSP_NUM_2, 0x00},
+	{CS43130_XSP_DEN_1, 0x02},
+	{CS43130_XSP_DEN_2, 0x00},
+	{CS43130_XSP_LRCK_HI_TIME_1, 0x1F},
+	{CS43130_XSP_LRCK_HI_TIME_2, 0x00},
+	{CS43130_XSP_LRCK_PERIOD_1, 0x3F},
+	{CS43130_XSP_LRCK_PERIOD_2, 0x00},
+	{CS43130_XSP_CLOCK_CONF, 0x0C},
+	{CS43130_XSP_FRAME_CONF, 0x0A},
+	{CS43130_ASP_CH_1_LOC, 0x00},
+	{CS43130_ASP_CH_2_LOC, 0x00},
+	{CS43130_ASP_CH_1_SZ_EN, 0x06},
+	{CS43130_ASP_CH_2_SZ_EN, 0x0E},
+	{CS43130_XSP_CH_1_LOC, 0x00},
+	{CS43130_XSP_CH_2_LOC, 0x00},
+	{CS43130_XSP_CH_1_SZ_EN, 0x06},
+	{CS43130_XSP_CH_2_SZ_EN, 0x0E},
+	{CS43130_DSD_VOL_B, 0x78},
+	{CS43130_DSD_VOL_A, 0x78},
+	{CS43130_DSD_PATH_CTL_1, 0xA8},
+	{CS43130_DSD_INT_CFG, 0x00},
+	{CS43130_DSD_PATH_CTL_2, 0x02},
+	{CS43130_DSD_PCM_MIX_CTL, 0x00},
+	{CS43130_DSD_PATH_CTL_3, 0x40},
+	{CS43130_HP_OUT_CTL_1, 0x30},
+	{CS43130_PCM_FILT_OPT, 0x02},
+	{CS43130_PCM_VOL_B, 0x78},
+	{CS43130_PCM_VOL_A, 0x78},
+	{CS43130_PCM_PATH_CTL_1, 0xA8},
+	{CS43130_PCM_PATH_CTL_2, 0x00},
+	{CS43130_CLASS_H_CTL, 0x1E},
+	{CS43130_HP_DETECT, 0x04},
+	{CS43130_HP_LOAD_1, 0x00},
+	{CS43130_HP_MEAS_LOAD_1, 0x00},
+	{CS43130_HP_MEAS_LOAD_2, 0x00},
+	{CS43130_INT_MASK_1, 0xFF},
+	{CS43130_INT_MASK_2, 0xFF},
+	{CS43130_INT_MASK_3, 0xFF},
+	{CS43130_INT_MASK_4, 0xFF},
+	{CS43130_INT_MASK_5, 0xFF},
+};
+
+static bool cs43130_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5:
+	case CS43130_HP_DC_STAT_1 ... CS43130_HP_DC_STAT_2:
+	case CS43130_HP_AC_STAT_1 ... CS43130_HP_AC_STAT_2:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs43130_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1:
+	case CS43130_SP_SRATE ... CS43130_PAD_INT_CFG:
+	case CS43130_PWDN_CTL:
+	case CS43130_CRYSTAL_SET:
+	case CS43130_PLL_SET_1 ... CS43130_PLL_SET_5:
+	case CS43130_PLL_SET_6:
+	case CS43130_PLL_SET_7:
+	case CS43130_PLL_SET_8:
+	case CS43130_PLL_SET_9:
+	case CS43130_PLL_SET_10:
+	case CS43130_CLKOUT_CTL:
+	case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF:
+	case CS43130_XSP_NUM_1 ... CS43130_XSP_FRAME_CONF:
+	case CS43130_ASP_CH_1_LOC:
+	case CS43130_ASP_CH_2_LOC:
+	case CS43130_ASP_CH_1_SZ_EN:
+	case CS43130_ASP_CH_2_SZ_EN:
+	case CS43130_XSP_CH_1_LOC:
+	case CS43130_XSP_CH_2_LOC:
+	case CS43130_XSP_CH_1_SZ_EN:
+	case CS43130_XSP_CH_2_SZ_EN:
+	case CS43130_DSD_VOL_B ... CS43130_DSD_PATH_CTL_3:
+	case CS43130_HP_OUT_CTL_1:
+	case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2:
+	case CS43130_CLASS_H_CTL:
+	case CS43130_HP_DETECT:
+	case CS43130_HP_STATUS:
+	case CS43130_HP_LOAD_1:
+	case CS43130_HP_MEAS_LOAD_1:
+	case CS43130_HP_MEAS_LOAD_2:
+	case CS43130_HP_DC_STAT_1:
+	case CS43130_HP_DC_STAT_2:
+	case CS43130_HP_AC_STAT_1:
+	case CS43130_HP_AC_STAT_2:
+	case CS43130_HP_LOAD_STAT:
+	case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5:
+	case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs43130_precious_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5:
+		return true;
+	default:
+		return false;
+	}
+}
+
+struct cs43130_pll_params {
+	unsigned int pll_in;
+	u8 sclk_prediv;
+	u8 pll_div_int;
+	u32 pll_div_frac;
+	u8 pll_mode;
+	u8 pll_divout;
+	unsigned int pll_out;
+	u8 pll_cal_ratio;
+};
+
+static const struct cs43130_pll_params pll_ratio_table[] = {
+	{9600000, 0x02, 0x49, 0x800000, 0x00, 0x08, 22579200, 151},
+	{9600000, 0x02, 0x50, 0x000000, 0x00, 0x08, 24576000, 164},
+
+	{11289600, 0x02, 0X40, 0, 0x01, 0x08, 22579200, 128},
+	{11289600, 0x02, 0x44, 0x06F700, 0x0, 0x08, 24576000, 139},
+
+	{12000000, 0x02, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120},
+	{12000000, 0x02, 0x40, 0x000000, 0x00, 0x08, 24576000, 131},
+
+	{12288000, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118},
+	{12288000, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128},
+
+	{13000000, 0x02, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111},
+	{13000000, 0x02, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121},
+
+	{19200000, 0x03, 0x49, 0x800000, 0x00, 0x08, 22579200, 151},
+	{19200000, 0x03, 0x50, 0x000000, 0x00, 0x08, 24576000, 164},
+
+	{22579200, 0, 0, 0, 0, 0, 22579200, 0},
+	{22579200, 0x03, 0x44, 0x06F700, 0x00, 0x08, 24576000, 139},
+
+	{24000000, 0x03, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120},
+	{24000000, 0x03, 0x40, 0x000000, 0x00, 0x08, 24576000, 131},
+
+	{24576000, 0x03, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118},
+	{24576000, 0, 0, 0, 0, 0, 24576000, 0},
+
+	{26000000, 0x03, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111},
+	{26000000, 0x03, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121},
+};
+
+static const struct cs43130_pll_params *cs43130_get_pll_table(
+		unsigned int freq_in, unsigned int freq_out)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
+		if (pll_ratio_table[i].pll_in == freq_in &&
+		    pll_ratio_table[i].pll_out == freq_out)
+			return &pll_ratio_table[i];
+	}
+
+	return NULL;
+}
+
+static int cs43130_pll_config(struct snd_soc_codec *codec)
+{
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+	const struct cs43130_pll_params *pll_entry;
+
+	dev_dbg(codec->dev, "cs43130->mclk = %u, cs43130->mclk_int = %u\n",
+		cs43130->mclk, cs43130->mclk_int);
+
+	pll_entry = cs43130_get_pll_table(cs43130->mclk, cs43130->mclk_int);
+	if (!pll_entry)
+		return -EINVAL;
+
+	if (pll_entry->pll_cal_ratio == 0) {
+		regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1,
+				   CS43130_PLL_START_MASK, 0);
+
+		cs43130->pll_bypass = true;
+		return 0;
+	}
+
+	cs43130->pll_bypass = false;
+
+	regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_2,
+			   CS43130_PLL_DIV_DATA_MASK,
+			   pll_entry->pll_div_frac >>
+			   CS43130_PLL_DIV_FRAC_0_DATA_SHIFT);
+	regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_3,
+			   CS43130_PLL_DIV_DATA_MASK,
+			   pll_entry->pll_div_frac >>
+			   CS43130_PLL_DIV_FRAC_1_DATA_SHIFT);
+	regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_4,
+			   CS43130_PLL_DIV_DATA_MASK,
+			   pll_entry->pll_div_frac >>
+			   CS43130_PLL_DIV_FRAC_2_DATA_SHIFT);
+	regmap_write(cs43130->regmap, CS43130_PLL_SET_5,
+		     pll_entry->pll_div_int);
+	regmap_write(cs43130->regmap, CS43130_PLL_SET_6, pll_entry->pll_divout);
+	regmap_write(cs43130->regmap, CS43130_PLL_SET_7,
+		     pll_entry->pll_cal_ratio);
+	regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_8,
+			   CS43130_PLL_MODE_MASK,
+			   pll_entry->pll_mode << CS43130_PLL_MODE_SHIFT);
+	regmap_write(cs43130->regmap, CS43130_PLL_SET_9,
+		     pll_entry->sclk_prediv);
+	regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1,
+			   CS43130_PLL_START_MASK, 1);
+
+	return 0;
+}
+
+static int cs43130_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+			   unsigned int freq_in, unsigned int freq_out)
+{
+	int ret = 0;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	switch (freq_in) {
+	case 9600000:
+	case 11289600:
+	case 12000000:
+	case 12288000:
+	case 13000000:
+	case 19200000:
+	case 22579200:
+	case 24000000:
+	case 24576000:
+	case 26000000:
+		cs43130->mclk = freq_in;
+		break;
+	default:
+		dev_err(codec->dev,
+			"unsupported pll input reference clock:%d\n", freq_in);
+		return -EINVAL;
+	}
+
+	switch (freq_out) {
+	case 22579200:
+		cs43130->mclk_int = freq_out;
+		break;
+	case 24576000:
+		cs43130->mclk_int = freq_out;
+		break;
+	default:
+		dev_err(codec->dev,
+			"unsupported pll output ref clock: %u\n", freq_out);
+		return -EINVAL;
+	}
+
+	ret = cs43130_pll_config(codec);
+	dev_dbg(codec->dev, "cs43130->pll_bypass = %d", cs43130->pll_bypass);
+	return ret;
+}
+
+static int cs43130_change_clksrc(struct snd_soc_codec *codec,
+				 enum cs43130_mclk_src_sel src)
+{
+	int ret;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+	int mclk_int_decoded;
+
+	if (src == cs43130->mclk_int_src) {
+		/* clk source has not changed */
+		return 0;
+	}
+
+	switch (cs43130->mclk_int) {
+	case CS43130_MCLK_22M:
+		mclk_int_decoded = CS43130_MCLK_22P5;
+		break;
+	case CS43130_MCLK_24M:
+		mclk_int_decoded = CS43130_MCLK_24P5;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid MCLK INT freq: %u\n", cs43130->mclk_int);
+		return -EINVAL;
+	}
+
+	switch (src) {
+	case CS43130_MCLK_SRC_EXT:
+		cs43130->pll_bypass = true;
+		cs43130->mclk_int_src = CS43130_MCLK_SRC_EXT;
+		if (cs43130->xtal_ibias == CS43130_XTAL_UNUSED) {
+			regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+					   CS43130_PDN_XTAL_MASK,
+					   1 << CS43130_PDN_XTAL_SHIFT);
+		} else {
+			reinit_completion(&cs43130->xtal_rdy);
+			regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+					   CS43130_XTAL_RDY_INT_MASK, 0);
+			regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+					   CS43130_PDN_XTAL_MASK, 0);
+			ret = wait_for_completion_timeout(&cs43130->xtal_rdy,
+							  msecs_to_jiffies(100));
+			regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+					   CS43130_XTAL_RDY_INT_MASK,
+					   1 << CS43130_XTAL_RDY_INT_SHIFT);
+			if (ret == 0) {
+				dev_err(codec->dev, "Timeout waiting for XTAL_READY interrupt\n");
+				return -ETIMEDOUT;
+			}
+		}
+
+		regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1,
+				   CS43130_MCLK_SRC_SEL_MASK,
+				   src << CS43130_MCLK_SRC_SEL_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1,
+				   CS43130_MCLK_INT_MASK,
+				   mclk_int_decoded << CS43130_MCLK_INT_SHIFT);
+		usleep_range(150, 200);
+
+		regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+				   CS43130_PDN_PLL_MASK,
+				   1 << CS43130_PDN_PLL_SHIFT);
+		break;
+	case CS43130_MCLK_SRC_PLL:
+		cs43130->pll_bypass = false;
+		cs43130->mclk_int_src = CS43130_MCLK_SRC_PLL;
+		if (cs43130->xtal_ibias == CS43130_XTAL_UNUSED) {
+			regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+					   CS43130_PDN_XTAL_MASK,
+					   1 << CS43130_PDN_XTAL_SHIFT);
+		} else {
+			reinit_completion(&cs43130->xtal_rdy);
+			regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+					   CS43130_XTAL_RDY_INT_MASK, 0);
+			regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+					   CS43130_PDN_XTAL_MASK, 0);
+			ret = wait_for_completion_timeout(&cs43130->xtal_rdy,
+							  msecs_to_jiffies(100));
+			regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+					   CS43130_XTAL_RDY_INT_MASK,
+					   1 << CS43130_XTAL_RDY_INT_SHIFT);
+			if (ret == 0) {
+				dev_err(codec->dev, "Timeout waiting for XTAL_READY interrupt\n");
+				return -ETIMEDOUT;
+			}
+		}
+
+		reinit_completion(&cs43130->pll_rdy);
+		regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+				   CS43130_PLL_RDY_INT_MASK, 0);
+		regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+				   CS43130_PDN_PLL_MASK, 0);
+		ret = wait_for_completion_timeout(&cs43130->pll_rdy,
+						  msecs_to_jiffies(100));
+		regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+				   CS43130_PLL_RDY_INT_MASK,
+				   1 << CS43130_PLL_RDY_INT_SHIFT);
+		if (ret == 0) {
+			dev_err(codec->dev, "Timeout waiting for PLL_READY interrupt\n");
+			return -ETIMEDOUT;
+		}
+
+		regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1,
+				   CS43130_MCLK_SRC_SEL_MASK,
+				   src << CS43130_MCLK_SRC_SEL_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1,
+				   CS43130_MCLK_INT_MASK,
+				   mclk_int_decoded << CS43130_MCLK_INT_SHIFT);
+		usleep_range(150, 200);
+		break;
+	case CS43130_MCLK_SRC_RCO:
+		cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO;
+
+		regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1,
+				   CS43130_MCLK_SRC_SEL_MASK,
+				   src << CS43130_MCLK_SRC_SEL_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1,
+				   CS43130_MCLK_INT_MASK,
+				   CS43130_MCLK_22P5 << CS43130_MCLK_INT_SHIFT);
+		usleep_range(150, 200);
+
+		regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+				   CS43130_PDN_XTAL_MASK,
+				   1 << CS43130_PDN_XTAL_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL,
+				   CS43130_PDN_PLL_MASK,
+				   1 << CS43130_PDN_PLL_SHIFT);
+		break;
+	default:
+		dev_err(codec->dev, "Invalid MCLK source value\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct cs43130_bitwidth_map cs43130_bitwidth_table[] = {
+	{8,	CS43130_SP_BIT_SIZE_8,	CS43130_CH_BIT_SIZE_8},
+	{16,	CS43130_SP_BIT_SIZE_16, CS43130_CH_BIT_SIZE_16},
+	{24,	CS43130_SP_BIT_SIZE_24, CS43130_CH_BIT_SIZE_24},
+	{32,	CS43130_SP_BIT_SIZE_32, CS43130_CH_BIT_SIZE_32},
+};
+
+static const struct cs43130_bitwidth_map *cs43130_get_bitwidth_table(
+				unsigned int bitwidth)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(cs43130_bitwidth_table); i++) {
+		if (cs43130_bitwidth_table[i].bitwidth == bitwidth)
+			return &cs43130_bitwidth_table[i];
+	}
+
+	return NULL;
+}
+
+static int cs43130_set_bitwidth(int dai_id, unsigned int bitwidth_dai,
+			  struct regmap *regmap)
+{
+	const struct cs43130_bitwidth_map *bw_map;
+
+	bw_map = cs43130_get_bitwidth_table(bitwidth_dai);
+	if (!bw_map)
+		return -EINVAL;
+
+	switch (dai_id) {
+	case CS43130_ASP_PCM_DAI:
+	case CS43130_ASP_DOP_DAI:
+		regmap_update_bits(regmap, CS43130_ASP_CH_1_SZ_EN,
+				   CS43130_CH_BITSIZE_MASK, bw_map->ch_bit);
+		regmap_update_bits(regmap, CS43130_ASP_CH_2_SZ_EN,
+				   CS43130_CH_BITSIZE_MASK, bw_map->ch_bit);
+		regmap_update_bits(regmap, CS43130_SP_BITSIZE,
+				   CS43130_ASP_BITSIZE_MASK, bw_map->sp_bit);
+		break;
+	case CS43130_XSP_DOP_DAI:
+		regmap_update_bits(regmap, CS43130_XSP_CH_1_SZ_EN,
+				   CS43130_CH_BITSIZE_MASK, bw_map->ch_bit);
+		regmap_update_bits(regmap, CS43130_XSP_CH_2_SZ_EN,
+				   CS43130_CH_BITSIZE_MASK, bw_map->ch_bit);
+		regmap_update_bits(regmap, CS43130_SP_BITSIZE,
+				   CS43130_XSP_BITSIZE_MASK, bw_map->sp_bit <<
+				   CS43130_XSP_BITSIZE_SHIFT);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct cs43130_rate_map cs43130_rate_table[] = {
+	{32000,		CS43130_ASP_SPRATE_32K},
+	{44100,		CS43130_ASP_SPRATE_44_1K},
+	{48000,		CS43130_ASP_SPRATE_48K},
+	{88200,		CS43130_ASP_SPRATE_88_2K},
+	{96000,		CS43130_ASP_SPRATE_96K},
+	{176400,	CS43130_ASP_SPRATE_176_4K},
+	{192000,	CS43130_ASP_SPRATE_192K},
+	{352800,	CS43130_ASP_SPRATE_352_8K},
+	{384000,	CS43130_ASP_SPRATE_384K},
+};
+
+static const struct cs43130_rate_map *cs43130_get_rate_table(int fs)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(cs43130_rate_table); i++) {
+		if (cs43130_rate_table[i].fs == fs)
+			return &cs43130_rate_table[i];
+	}
+
+	return NULL;
+}
+
+static const struct cs43130_clk_gen *cs43130_get_clk_gen(int mclk_int, int fs,
+		const struct cs43130_clk_gen *clk_gen_table, int len_clk_gen_table)
+{
+	int i;
+
+	for (i = 0; i < len_clk_gen_table; i++) {
+		if (clk_gen_table[i].mclk_int == mclk_int &&
+		    clk_gen_table[i].fs == fs)
+			return &clk_gen_table[i];
+	}
+
+	return NULL;
+}
+
+static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk,
+			      struct snd_pcm_hw_params *params,
+			      struct cs43130_private *cs43130)
+{
+	u16 frm_size;
+	u16 hi_size;
+	u8 frm_delay;
+	u8 frm_phase;
+	u8 frm_data;
+	u8 sclk_edge;
+	u8 lrck_edge;
+	u8 clk_data;
+	u8 loc_ch1;
+	u8 loc_ch2;
+	u8 dai_mode_val;
+	const struct cs43130_clk_gen *clk_gen;
+
+	switch (cs43130->dais[dai_id].dai_format) {
+	case SND_SOC_DAIFMT_I2S:
+		hi_size = bitwidth_sclk;
+		frm_delay = 2;
+		frm_phase = 0;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		hi_size = bitwidth_sclk;
+		frm_delay = 2;
+		frm_phase = 1;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		hi_size = 1;
+		frm_delay = 2;
+		frm_phase = 1;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		hi_size = 1;
+		frm_delay = 0;
+		frm_phase = 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (cs43130->dais[dai_id].dai_mode) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		dai_mode_val = 0;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		dai_mode_val = 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	frm_size = bitwidth_sclk * params_channels(params);
+	sclk_edge = 1;
+	lrck_edge = 0;
+	loc_ch1 = 0;
+	loc_ch2 = bitwidth_sclk * (params_channels(params) - 1);
+
+	frm_data = frm_delay & CS43130_SP_FSD_MASK;
+	frm_data |= (frm_phase << CS43130_SP_STP_SHIFT) & CS43130_SP_STP_MASK;
+
+	clk_data = lrck_edge & CS43130_SP_LCPOL_IN_MASK;
+	clk_data |= (lrck_edge << CS43130_SP_LCPOL_OUT_SHIFT) &
+		    CS43130_SP_LCPOL_OUT_MASK;
+	clk_data |= (sclk_edge << CS43130_SP_SCPOL_IN_SHIFT) &
+		    CS43130_SP_SCPOL_IN_MASK;
+	clk_data |= (sclk_edge << CS43130_SP_SCPOL_OUT_SHIFT) &
+		    CS43130_SP_SCPOL_OUT_MASK;
+	clk_data |= (dai_mode_val << CS43130_SP_MODE_SHIFT) &
+		    CS43130_SP_MODE_MASK;
+
+	switch (dai_id) {
+	case CS43130_ASP_PCM_DAI:
+	case CS43130_ASP_DOP_DAI:
+		regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_1,
+			CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >>
+			CS43130_SP_LCPR_LSB_DATA_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_2,
+			CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >>
+			CS43130_SP_LCPR_MSB_DATA_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_1,
+			CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >>
+			CS43130_SP_LCHI_LSB_DATA_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_2,
+			CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >>
+			CS43130_SP_LCHI_MSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_ASP_FRAME_CONF, frm_data);
+		regmap_write(cs43130->regmap, CS43130_ASP_CH_1_LOC, loc_ch1);
+		regmap_write(cs43130->regmap, CS43130_ASP_CH_2_LOC, loc_ch2);
+		regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_1_SZ_EN,
+			CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_2_SZ_EN,
+			CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_ASP_CLOCK_CONF, clk_data);
+		break;
+	case CS43130_XSP_DOP_DAI:
+		regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_PERIOD_1,
+			CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >>
+			CS43130_SP_LCPR_LSB_DATA_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_PERIOD_2,
+			CS43130_SP_LCPR_DATA_MASK, (frm_size - 1) >>
+			CS43130_SP_LCPR_MSB_DATA_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_HI_TIME_1,
+			CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >>
+			CS43130_SP_LCHI_LSB_DATA_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_XSP_LRCK_HI_TIME_2,
+			CS43130_SP_LCHI_DATA_MASK, (hi_size - 1) >>
+			CS43130_SP_LCHI_MSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_XSP_FRAME_CONF, frm_data);
+		regmap_write(cs43130->regmap, CS43130_XSP_CH_1_LOC, loc_ch1);
+		regmap_write(cs43130->regmap, CS43130_XSP_CH_2_LOC, loc_ch2);
+		regmap_update_bits(cs43130->regmap, CS43130_XSP_CH_1_SZ_EN,
+			CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_XSP_CH_2_SZ_EN,
+			CS43130_CH_EN_MASK, 1 << CS43130_CH_EN_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_XSP_CLOCK_CONF, clk_data);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (frm_size) {
+	case 16:
+		clk_gen = cs43130_get_clk_gen(cs43130->mclk_int,
+					      params_rate(params),
+					      cs43130_16_clk_gen,
+					      ARRAY_SIZE(cs43130_16_clk_gen));
+		break;
+	case 32:
+		clk_gen = cs43130_get_clk_gen(cs43130->mclk_int,
+					      params_rate(params),
+					      cs43130_32_clk_gen,
+					      ARRAY_SIZE(cs43130_32_clk_gen));
+		break;
+	case 48:
+		clk_gen = cs43130_get_clk_gen(cs43130->mclk_int,
+					      params_rate(params),
+					      cs43130_48_clk_gen,
+					      ARRAY_SIZE(cs43130_48_clk_gen));
+		break;
+	case 64:
+		clk_gen = cs43130_get_clk_gen(cs43130->mclk_int,
+					      params_rate(params),
+					      cs43130_64_clk_gen,
+					      ARRAY_SIZE(cs43130_64_clk_gen));
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (!clk_gen)
+		return -EINVAL;
+
+	switch (dai_id) {
+	case CS43130_ASP_PCM_DAI:
+	case CS43130_ASP_DOP_DAI:
+		regmap_write(cs43130->regmap, CS43130_ASP_DEN_1,
+			     (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >>
+			     CS43130_SP_M_LSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_ASP_DEN_2,
+			     (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >>
+			     CS43130_SP_M_MSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_ASP_NUM_1,
+			     (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >>
+			     CS43130_SP_N_LSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_ASP_NUM_2,
+			     (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >>
+			     CS43130_SP_N_MSB_DATA_SHIFT);
+		break;
+	case CS43130_XSP_DOP_DAI:
+		regmap_write(cs43130->regmap, CS43130_XSP_DEN_1,
+			     (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >>
+			     CS43130_SP_M_LSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_XSP_DEN_2,
+			     (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >>
+			     CS43130_SP_M_MSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_XSP_NUM_1,
+			     (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >>
+			     CS43130_SP_N_LSB_DATA_SHIFT);
+		regmap_write(cs43130->regmap, CS43130_XSP_NUM_2,
+			     (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >>
+			     CS43130_SP_N_MSB_DATA_SHIFT);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int cs43130_pcm_dsd_mix(bool en, struct regmap *regmap)
+{
+	if (en) {
+		regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL,
+				   CS43130_MIX_PCM_PREP_MASK,
+				   1 << CS43130_MIX_PCM_PREP_SHIFT);
+		usleep_range(6000, 6050);
+		regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL,
+				   CS43130_MIX_PCM_DSD_MASK,
+				   1 << CS43130_MIX_PCM_DSD_SHIFT);
+	} else {
+		regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL,
+				   CS43130_MIX_PCM_DSD_MASK,
+				   0 << CS43130_MIX_PCM_DSD_SHIFT);
+		usleep_range(1600, 1650);
+		regmap_update_bits(regmap, CS43130_DSD_PCM_MIX_CTL,
+				   CS43130_MIX_PCM_PREP_MASK,
+				   0 << CS43130_MIX_PCM_PREP_SHIFT);
+	}
+
+	return 0;
+}
+
+static int cs43130_dsd_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+	unsigned int required_clk;
+	u8 dsd_speed;
+
+	mutex_lock(&cs43130->clk_mutex);
+	if (!cs43130->clk_req) {
+		/* no DAI is currently using clk */
+		if (!(CS43130_MCLK_22M % params_rate(params)))
+			required_clk = CS43130_MCLK_22M;
+		else
+			required_clk = CS43130_MCLK_24M;
+
+		cs43130_set_pll(codec, 0, 0, cs43130->mclk, required_clk);
+		if (cs43130->pll_bypass)
+			cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT);
+		else
+			cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL);
+	}
+
+	cs43130->clk_req++;
+	if (cs43130->clk_req == 2)
+		cs43130_pcm_dsd_mix(true, cs43130->regmap);
+	mutex_unlock(&cs43130->clk_mutex);
+
+	switch (params_rate(params)) {
+	case 176400:
+		dsd_speed = 0;
+		break;
+	case 352800:
+		dsd_speed = 1;
+		break;
+	default:
+		dev_err(codec->dev, "Rate(%u) not supported\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	if (cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM)
+		regmap_update_bits(cs43130->regmap, CS43130_DSD_INT_CFG,
+				   CS43130_DSD_MASTER, CS43130_DSD_MASTER);
+	else
+		regmap_update_bits(cs43130->regmap, CS43130_DSD_INT_CFG,
+				   CS43130_DSD_MASTER, 0);
+
+	regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2,
+			   CS43130_DSD_SPEED_MASK,
+			   dsd_speed << CS43130_DSD_SPEED_SHIFT);
+	regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2,
+			   CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_DSD <<
+			   CS43130_DSD_SRC_SHIFT);
+
+	return 0;
+}
+
+static int cs43130_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+	const struct cs43130_rate_map *rate_map;
+	unsigned int sclk = cs43130->dais[dai->id].sclk;
+	unsigned int bitwidth_sclk;
+	unsigned int bitwidth_dai = (unsigned int)(params_width(params));
+	unsigned int required_clk;
+	u8 dsd_speed;
+
+	mutex_lock(&cs43130->clk_mutex);
+	if (!cs43130->clk_req) {
+		/* no DAI is currently using clk */
+		if (!(CS43130_MCLK_22M % params_rate(params)))
+			required_clk = CS43130_MCLK_22M;
+		else
+			required_clk = CS43130_MCLK_24M;
+
+		cs43130_set_pll(codec, 0, 0, cs43130->mclk, required_clk);
+		if (cs43130->pll_bypass)
+			cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT);
+		else
+			cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL);
+	}
+
+	cs43130->clk_req++;
+	if (cs43130->clk_req == 2)
+		cs43130_pcm_dsd_mix(true, cs43130->regmap);
+	mutex_unlock(&cs43130->clk_mutex);
+
+	switch (dai->id) {
+	case CS43130_ASP_DOP_DAI:
+	case CS43130_XSP_DOP_DAI:
+		/* DoP bitwidth is always 24-bit */
+		bitwidth_dai = 24;
+		sclk = params_rate(params) * bitwidth_dai *
+		       params_channels(params);
+
+		switch (params_rate(params)) {
+		case 176400:
+			dsd_speed = 0;
+			break;
+		case 352800:
+			dsd_speed = 1;
+			break;
+		default:
+			dev_err(codec->dev, "Rate(%u) not supported\n",
+				params_rate(params));
+			return -EINVAL;
+		}
+
+		regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2,
+				   CS43130_DSD_SPEED_MASK,
+				   dsd_speed << CS43130_DSD_SPEED_SHIFT);
+		break;
+	case CS43130_ASP_PCM_DAI:
+		rate_map = cs43130_get_rate_table(params_rate(params));
+		if (!rate_map)
+			return -EINVAL;
+
+		regmap_write(cs43130->regmap, CS43130_SP_SRATE, rate_map->val);
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI (%d)\n", dai->id);
+		return -EINVAL;
+	}
+
+	switch (dai->id) {
+	case CS43130_ASP_DOP_DAI:
+		regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2,
+				   CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_ASP <<
+				   CS43130_DSD_SRC_SHIFT);
+		break;
+	case CS43130_XSP_DOP_DAI:
+		regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2,
+				   CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_XSP <<
+				   CS43130_DSD_SRC_SHIFT);
+	}
+
+	if (!sclk && cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM)
+		/* Calculate SCLK in master mode if unassigned */
+		sclk = params_rate(params) * bitwidth_dai *
+		       params_channels(params);
+
+	if (!sclk) {
+		/* at this point, SCLK must be set */
+		dev_err(codec->dev, "SCLK freq is not set\n");
+		return -EINVAL;
+	}
+
+	bitwidth_sclk = (sclk / params_rate(params)) / params_channels(params);
+	if (bitwidth_sclk < bitwidth_dai) {
+		dev_err(codec->dev, "Format not supported: SCLK freq is too low\n");
+		return -EINVAL;
+	}
+
+	dev_dbg(codec->dev,
+		"sclk = %u, fs = %d, bitwidth_dai = %u\n",
+		sclk, params_rate(params), bitwidth_dai);
+
+	dev_dbg(codec->dev,
+		"bitwidth_sclk = %u, num_ch = %u\n",
+		bitwidth_sclk, params_channels(params));
+
+	cs43130_set_bitwidth(dai->id, bitwidth_dai, cs43130->regmap);
+	cs43130_set_sp_fmt(dai->id, bitwidth_sclk, params, cs43130);
+
+	return 0;
+}
+
+static int cs43130_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	mutex_lock(&cs43130->clk_mutex);
+	cs43130->clk_req--;
+	if (!cs43130->clk_req) {
+		/* no DAI is currently using clk */
+		cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO);
+		cs43130_pcm_dsd_mix(false, cs43130->regmap);
+	}
+	mutex_unlock(&cs43130->clk_mutex);
+
+	return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(pcm_vol_tlv, -12750, 50, 1);
+
+static const char * const pcm_ch_text[] = {
+	"Left-Right Ch",
+	"Left-Left Ch",
+	"Right-Left Ch",
+	"Right-Right Ch",
+};
+
+static const struct reg_sequence pcm_ch_en_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{0x180005, 0x8C},
+	{0x180007, 0xAB},
+	{0x180015, 0x31},
+	{0x180017, 0xB2},
+	{0x180025, 0x30},
+	{0x180027, 0x84},
+	{0x180035, 0x9C},
+	{0x180037, 0xAE},
+	{0x18000D, 0x24},
+	{0x18000F, 0xA3},
+	{0x18001D, 0x05},
+	{0x18001F, 0xD4},
+	{0x18002D, 0x0B},
+	{0x18002F, 0xC7},
+	{0x18003D, 0x71},
+	{0x18003F, 0xE7},
+	{CS43130_DXD1, 0},
+};
+
+static const struct reg_sequence pcm_ch_dis_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{0x180005, 0x24},
+	{0x180007, 0xA3},
+	{0x180015, 0x05},
+	{0x180017, 0xD4},
+	{0x180025, 0x0B},
+	{0x180027, 0xC7},
+	{0x180035, 0x71},
+	{0x180037, 0xE7},
+	{0x18000D, 0x8C},
+	{0x18000F, 0xAB},
+	{0x18001D, 0x31},
+	{0x18001F, 0xB2},
+	{0x18002D, 0x30},
+	{0x18002F, 0x84},
+	{0x18003D, 0x9C},
+	{0x18003F, 0xAE},
+	{CS43130_DXD1, 0},
+};
+
+static int cs43130_pcm_ch_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	return snd_soc_get_enum_double(kcontrol, ucontrol);
+}
+
+static int cs43130_pcm_ch_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int *item = ucontrol->value.enumerated.item;
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+
+	if (item[0] >= e->items)
+		return -EINVAL;
+	val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l;
+
+	switch (cs43130->dev_id) {
+	case CS43131_CHIP_ID:
+	case CS43198_CHIP_ID:
+		if (val >= 2)
+			regmap_multi_reg_write(cs43130->regmap, pcm_ch_en_seq,
+					       ARRAY_SIZE(pcm_ch_en_seq));
+		else
+			regmap_multi_reg_write(cs43130->regmap, pcm_ch_dis_seq,
+					       ARRAY_SIZE(pcm_ch_dis_seq));
+	}
+
+	return snd_soc_put_enum_double(kcontrol, ucontrol);
+}
+
+static SOC_ENUM_SINGLE_DECL(pcm_ch_enum, CS43130_PCM_PATH_CTL_2, 0,
+			    pcm_ch_text);
+
+static const char * const pcm_spd_texts[] = {
+	"Fast",
+	"Slow",
+};
+
+static SOC_ENUM_SINGLE_DECL(pcm_spd_enum, CS43130_PCM_FILT_OPT, 7,
+			    pcm_spd_texts);
+
+static const char * const dsd_texts[] = {
+	"Off",
+	"BCKA Mode",
+	"BCKD Mode",
+};
+
+static const unsigned int dsd_values[] = {
+	CS43130_DSD_SRC_DSD,
+	CS43130_DSD_SRC_ASP,
+	CS43130_DSD_SRC_XSP,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(dsd_enum, CS43130_DSD_INT_CFG, 0, 0x03,
+				  dsd_texts, dsd_values);
+
+static const struct snd_kcontrol_new cs43130_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Master Playback Volume",
+			 CS43130_PCM_VOL_A, CS43130_PCM_VOL_B, 0, 0xFF, 1,
+			 pcm_vol_tlv),
+	SOC_DOUBLE_R_TLV("Master DSD Playback Volume",
+			 CS43130_DSD_VOL_A, CS43130_DSD_VOL_B, 0, 0xFF, 1,
+			 pcm_vol_tlv),
+	SOC_ENUM_EXT("PCM Ch Select", pcm_ch_enum, cs43130_pcm_ch_get,
+		     cs43130_pcm_ch_put),
+	SOC_ENUM("PCM Filter Speed", pcm_spd_enum),
+	SOC_SINGLE("PCM Phase Compensation", CS43130_PCM_FILT_OPT, 6, 1, 0),
+	SOC_SINGLE("PCM Nonoversample Emulate", CS43130_PCM_FILT_OPT, 5, 1, 0),
+	SOC_SINGLE("PCM High-pass Filter", CS43130_PCM_FILT_OPT, 1, 1, 0),
+	SOC_SINGLE("PCM De-emphasis Filter", CS43130_PCM_FILT_OPT, 0, 1, 0),
+	SOC_ENUM("DSD Phase Modulation", dsd_enum),
+};
+
+static const struct reg_sequence pcm_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD7, 0x01},
+	{CS43130_DXD8, 0},
+	{CS43130_DXD9, 0x01},
+	{CS43130_DXD3, 0x12},
+	{CS43130_DXD4, 0},
+	{CS43130_DXD10, 0x28},
+	{CS43130_DXD11, 0x28},
+	{CS43130_DXD1, 0},
+};
+
+static const struct reg_sequence dsd_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD7, 0x01},
+	{CS43130_DXD8, 0},
+	{CS43130_DXD9, 0x01},
+	{CS43130_DXD3, 0x12},
+	{CS43130_DXD4, 0},
+	{CS43130_DXD10, 0x1E},
+	{CS43130_DXD11, 0x20},
+	{CS43130_DXD1, 0},
+};
+
+static const struct reg_sequence pop_free_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD12, 0x0A},
+	{CS43130_DXD1, 0},
+};
+
+static const struct reg_sequence pop_free_seq2[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD13, 0x20},
+	{CS43130_DXD1, 0},
+};
+
+static const struct reg_sequence mute_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD3, 0x12},
+	{CS43130_DXD5, 0x02},
+	{CS43130_DXD4, 0x12},
+	{CS43130_DXD1, 0},
+};
+
+static const struct reg_sequence unmute_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD3, 0x10},
+	{CS43130_DXD5, 0},
+	{CS43130_DXD4, 0x16},
+	{CS43130_DXD1, 0},
+};
+
+static int cs43130_dsd_event(struct snd_soc_dapm_widget *w,
+			      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, dsd_seq,
+					       ARRAY_SIZE(dsd_seq));
+		}
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_1,
+				   CS43130_MUTE_MASK, 0);
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, unmute_seq,
+					       ARRAY_SIZE(unmute_seq));
+		}
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, mute_seq,
+					       ARRAY_SIZE(mute_seq));
+			regmap_update_bits(cs43130->regmap,
+					   CS43130_DSD_PATH_CTL_1,
+					   CS43130_MUTE_MASK, CS43130_MUTE_EN);
+			/*
+			 * DSD Power Down Sequence
+			 * According to Design, 130ms is preferred.
+			 */
+			msleep(130);
+			break;
+		case CS43131_CHIP_ID:
+		case CS43198_CHIP_ID:
+			regmap_update_bits(cs43130->regmap,
+					   CS43130_DSD_PATH_CTL_1,
+					   CS43130_MUTE_MASK, CS43130_MUTE_EN);
+		}
+		break;
+	default:
+		dev_err(codec->dev, "Invalid event = 0x%x\n", event);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int cs43130_pcm_event(struct snd_soc_dapm_widget *w,
+			      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, pcm_seq,
+					       ARRAY_SIZE(pcm_seq));
+		}
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		regmap_update_bits(cs43130->regmap, CS43130_PCM_PATH_CTL_1,
+				   CS43130_MUTE_MASK, 0);
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, unmute_seq,
+					       ARRAY_SIZE(unmute_seq));
+		}
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, mute_seq,
+					       ARRAY_SIZE(mute_seq));
+			regmap_update_bits(cs43130->regmap,
+					   CS43130_PCM_PATH_CTL_1,
+					   CS43130_MUTE_MASK, CS43130_MUTE_EN);
+			/*
+			 * PCM Power Down Sequence
+			 * According to Design, 130ms is preferred.
+			 */
+			msleep(130);
+			break;
+		case CS43131_CHIP_ID:
+		case CS43198_CHIP_ID:
+			regmap_update_bits(cs43130->regmap,
+					   CS43130_PCM_PATH_CTL_1,
+					   CS43130_MUTE_MASK, CS43130_MUTE_EN);
+		}
+		break;
+	default:
+		dev_err(codec->dev, "Invalid event = 0x%x\n", event);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static const struct reg_sequence dac_postpmu_seq[] = {
+	{CS43130_DXD9, 0x0C},
+	{CS43130_DXD3, 0x10},
+	{CS43130_DXD4, 0x20},
+};
+
+static const struct reg_sequence dac_postpmd_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD6, 0x01},
+	{CS43130_DXD1, 0},
+};
+
+static int cs43130_dac_event(struct snd_soc_dapm_widget *w,
+			     struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, pop_free_seq,
+					       ARRAY_SIZE(pop_free_seq));
+			break;
+		case CS43131_CHIP_ID:
+		case CS43198_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, pop_free_seq2,
+					       ARRAY_SIZE(pop_free_seq2));
+		}
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		usleep_range(10000, 10050);
+
+		regmap_write(cs43130->regmap, CS43130_DXD1, 0x99);
+
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, dac_postpmu_seq,
+					       ARRAY_SIZE(dac_postpmu_seq));
+			/*
+			 * Per datasheet, Sec. PCM Power-Up Sequence.
+			 * According to Design, CS43130_DXD12 must be 0 to meet
+			 * THDN and Dynamic Range spec.
+			 */
+			msleep(1000);
+			regmap_write(cs43130->regmap, CS43130_DXD12, 0);
+			break;
+		case CS43131_CHIP_ID:
+		case CS43198_CHIP_ID:
+			usleep_range(12000, 12010);
+			regmap_write(cs43130->regmap, CS43130_DXD13, 0);
+		}
+
+		regmap_write(cs43130->regmap, CS43130_DXD1, 0);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		switch (cs43130->dev_id) {
+		case CS43130_CHIP_ID:
+		case CS4399_CHIP_ID:
+			regmap_multi_reg_write(cs43130->regmap, dac_postpmd_seq,
+					       ARRAY_SIZE(dac_postpmd_seq));
+		}
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAC event = 0x%x\n", event);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static const struct reg_sequence hpin_prepmd_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD15, 0x64},
+	{CS43130_DXD14, 0},
+	{CS43130_DXD2, 0},
+	{CS43130_DXD1, 0},
+};
+
+static const struct reg_sequence hpin_postpmu_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD2, 1},
+	{CS43130_DXD14, 0xDC},
+	{CS43130_DXD15, 0xE4},
+	{CS43130_DXD1, 0},
+};
+
+static int cs43130_hpin_event(struct snd_soc_dapm_widget *w,
+			      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMD:
+		regmap_multi_reg_write(cs43130->regmap, hpin_prepmd_seq,
+				       ARRAY_SIZE(hpin_prepmd_seq));
+		break;
+	case SND_SOC_DAPM_PRE_PMU:
+		regmap_multi_reg_write(cs43130->regmap, hpin_postpmu_seq,
+				       ARRAY_SIZE(hpin_postpmu_seq));
+		break;
+	default:
+		dev_err(codec->dev, "Invalid HPIN event = 0x%x\n", event);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget digital_hp_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("HPOUTA"),
+	SND_SOC_DAPM_OUTPUT("HPOUTB"),
+
+	SND_SOC_DAPM_AIF_IN_E("ASPIN PCM", NULL, 0, CS43130_PWDN_CTL,
+			      CS43130_PDN_ASP_SHIFT, 1, cs43130_pcm_event,
+			      (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_PRE_PMD)),
+
+	SND_SOC_DAPM_AIF_IN_E("ASPIN DoP", NULL, 0, CS43130_PWDN_CTL,
+			      CS43130_PDN_ASP_SHIFT, 1, cs43130_dsd_event,
+			      (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_PRE_PMD)),
+
+	SND_SOC_DAPM_AIF_IN_E("XSPIN DoP", NULL, 0, CS43130_PWDN_CTL,
+			      CS43130_PDN_XSP_SHIFT, 1, cs43130_dsd_event,
+			      (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_PRE_PMD)),
+
+	SND_SOC_DAPM_AIF_IN_E("XSPIN DSD", NULL, 0, CS43130_PWDN_CTL,
+			      CS43130_PDN_DSDIF_SHIFT, 1, cs43130_dsd_event,
+			      (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_PRE_PMD)),
+
+	SND_SOC_DAPM_DAC("DSD", NULL, CS43130_DSD_PATH_CTL_2,
+			 CS43130_DSD_EN_SHIFT, 0),
+
+	SND_SOC_DAPM_DAC_E("HiFi DAC", NULL, CS43130_PWDN_CTL,
+			   CS43130_PDN_HP_SHIFT, 1, cs43130_dac_event,
+			   (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			    SND_SOC_DAPM_POST_PMD)),
+};
+
+static const struct snd_soc_dapm_widget analog_hp_widgets[] = {
+	SND_SOC_DAPM_DAC_E("Analog Playback", NULL, CS43130_HP_OUT_CTL_1,
+			   CS43130_HP_IN_EN_SHIFT, 0, cs43130_hpin_event,
+			   (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)),
+};
+
+static struct snd_soc_dapm_widget all_hp_widgets[
+			ARRAY_SIZE(digital_hp_widgets) +
+			ARRAY_SIZE(analog_hp_widgets)];
+
+static const struct snd_soc_dapm_route digital_hp_routes[] = {
+	{"ASPIN PCM", NULL, "ASP PCM Playback"},
+	{"ASPIN DoP", NULL, "ASP DoP Playback"},
+	{"XSPIN DoP", NULL, "XSP DoP Playback"},
+	{"XSPIN DSD", NULL, "XSP DSD Playback"},
+	{"DSD", NULL, "ASPIN DoP"},
+	{"DSD", NULL, "XSPIN DoP"},
+	{"DSD", NULL, "XSPIN DSD"},
+	{"HiFi DAC", NULL, "ASPIN PCM"},
+	{"HiFi DAC", NULL, "DSD"},
+	{"HPOUTA", NULL, "HiFi DAC"},
+	{"HPOUTB", NULL, "HiFi DAC"},
+};
+
+static const struct snd_soc_dapm_route analog_hp_routes[] = {
+	{"HPOUTA", NULL, "Analog Playback"},
+	{"HPOUTB", NULL, "Analog Playback"},
+};
+
+static struct snd_soc_dapm_route all_hp_routes[
+			ARRAY_SIZE(digital_hp_routes) +
+			ARRAY_SIZE(analog_hp_routes)];
+
+static const unsigned int cs43130_asp_src_rates[] = {
+	32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000
+};
+
+static const struct snd_pcm_hw_constraint_list cs43130_asp_constraints = {
+	.count	= ARRAY_SIZE(cs43130_asp_src_rates),
+	.list	= cs43130_asp_src_rates,
+};
+
+static int cs43130_pcm_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_RATE,
+					  &cs43130_asp_constraints);
+}
+
+static const unsigned int cs43130_dop_src_rates[] = {
+	176400, 352800,
+};
+
+static const struct snd_pcm_hw_constraint_list cs43130_dop_constraints = {
+	.count	= ARRAY_SIZE(cs43130_dop_src_rates),
+	.list	= cs43130_dop_src_rates,
+};
+
+static int cs43130_dop_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_RATE,
+					  &cs43130_dop_constraints);
+}
+
+static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBS_CFS;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported mode\n");
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_DSP_A;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_DSP_B;
+		break;
+	default:
+		dev_err(codec->dev,
+			"unsupported audio format\n");
+		return -EINVAL;
+	}
+
+	dev_dbg(codec->dev, "dai_id = %d,  dai_mode = %u, dai_format = %u\n",
+		codec_dai->id,
+		cs43130->dais[codec_dai->id].dai_mode,
+		cs43130->dais[codec_dai->id].dai_format);
+
+	return 0;
+}
+
+static int cs43130_dsd_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBS_CFS;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported DAI format.\n");
+		return -EINVAL;
+	}
+
+	dev_dbg(codec->dev, "dai_mode = 0x%x\n",
+		cs43130->dais[codec_dai->id].dai_mode);
+
+	return 0;
+}
+
+static int cs43130_set_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	cs43130->dais[codec_dai->id].sclk = freq;
+	dev_dbg(codec->dev, "dai_id = %d,  sclk = %u\n", codec_dai->id,
+		cs43130->dais[codec_dai->id].sclk);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops cs43130_pcm_ops = {
+	.startup	= cs43130_pcm_startup,
+	.hw_params	= cs43130_hw_params,
+	.hw_free	= cs43130_hw_free,
+	.set_sysclk	= cs43130_set_sysclk,
+	.set_fmt	= cs43130_pcm_set_fmt,
+};
+
+static const struct snd_soc_dai_ops cs43130_dop_ops = {
+	.startup	= cs43130_dop_startup,
+	.hw_params	= cs43130_hw_params,
+	.hw_free	= cs43130_hw_free,
+	.set_sysclk	= cs43130_set_sysclk,
+	.set_fmt	= cs43130_pcm_set_fmt,
+};
+
+static const struct snd_soc_dai_ops cs43130_dsd_ops = {
+	.startup        = cs43130_dop_startup,
+	.hw_params	= cs43130_dsd_hw_params,
+	.hw_free	= cs43130_hw_free,
+	.set_fmt	= cs43130_dsd_set_fmt,
+};
+
+static struct snd_soc_dai_driver cs43130_dai[] = {
+	{
+		.name = "cs43130-asp-pcm",
+		.id = CS43130_ASP_PCM_DAI,
+		.playback = {
+			.stream_name = "ASP PCM Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = CS43130_PCM_FORMATS,
+		},
+		.ops = &cs43130_pcm_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "cs43130-asp-dop",
+		.id = CS43130_ASP_DOP_DAI,
+		.playback = {
+			.stream_name = "ASP DoP Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = CS43130_DOP_FORMATS,
+		},
+		.ops = &cs43130_dop_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "cs43130-xsp-dop",
+		.id = CS43130_XSP_DOP_DAI,
+		.playback = {
+			.stream_name = "XSP DoP Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = CS43130_DOP_FORMATS,
+		},
+		.ops = &cs43130_dop_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "cs43130-xsp-dsd",
+		.id = CS43130_XSP_DSD_DAI,
+		.playback = {
+			.stream_name = "XSP DSD Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = CS43130_DOP_FORMATS,
+		},
+		.ops = &cs43130_dsd_ops,
+	},
+
+};
+
+static int cs43130_codec_set_sysclk(struct snd_soc_codec *codec,
+				    int clk_id, int source, unsigned int freq,
+				    int dir)
+{
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+
+	dev_dbg(codec->dev, "clk_id = %d, source = %d, freq = %d, dir = %d\n",
+		clk_id, source, freq, dir);
+
+	switch (freq) {
+	case CS43130_MCLK_22M:
+	case CS43130_MCLK_24M:
+		cs43130->mclk = freq;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid MCLK INT freq: %u\n", freq);
+		return -EINVAL;
+	}
+
+	if (source == CS43130_MCLK_SRC_EXT) {
+		cs43130->pll_bypass = true;
+	} else {
+		dev_err(codec->dev, "Invalid MCLK source\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static inline u16 cs43130_get_ac_reg_val(u16 ac_freq)
+{
+	/* AC freq is counted in 5.94Hz step. */
+	return ac_freq / 6;
+}
+
+static int cs43130_show_dc(struct device *dev, char *buf, u8 ch)
+{
+	struct i2c_client *client = to_i2c_client(dev);
+	struct cs43130_private *cs43130 = i2c_get_clientdata(client);
+
+	if (!cs43130->hpload_done)
+		return scnprintf(buf, PAGE_SIZE, "NO_HPLOAD\n");
+	else
+		return scnprintf(buf, PAGE_SIZE, "%u\n",
+				 cs43130->hpload_dc[ch]);
+}
+
+static ssize_t cs43130_show_dc_l(struct device *dev,
+				 struct device_attribute *attr, char *buf)
+{
+	return cs43130_show_dc(dev, buf, HP_LEFT);
+}
+
+static ssize_t cs43130_show_dc_r(struct device *dev,
+				 struct device_attribute *attr, char *buf)
+{
+	return cs43130_show_dc(dev, buf, HP_RIGHT);
+}
+
+static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = {
+	24,
+	43,
+	93,
+	200,
+	431,
+	928,
+	2000,
+	4309,
+	9283,
+	20000,
+};
+
+static int cs43130_show_ac(struct device *dev, char *buf, u8 ch)
+{
+	int i, j = 0, tmp;
+	struct i2c_client *client = to_i2c_client(dev);
+	struct cs43130_private *cs43130 = i2c_get_clientdata(client);
+
+	if (cs43130->hpload_done && cs43130->ac_meas) {
+		for (i = 0; i < ARRAY_SIZE(cs43130_ac_freq); i++) {
+			tmp = scnprintf(buf + j, PAGE_SIZE - j, "%u\n",
+					cs43130->hpload_ac[i][ch]);
+			if (!tmp)
+				break;
+
+			j += tmp;
+		}
+
+		return j;
+	} else {
+		return scnprintf(buf, PAGE_SIZE, "NO_HPLOAD\n");
+	}
+}
+
+static ssize_t cs43130_show_ac_l(struct device *dev,
+				 struct device_attribute *attr, char *buf)
+{
+	return cs43130_show_ac(dev, buf, HP_LEFT);
+}
+
+static ssize_t cs43130_show_ac_r(struct device *dev,
+				 struct device_attribute *attr, char *buf)
+{
+	return cs43130_show_ac(dev, buf, HP_RIGHT);
+}
+
+static DEVICE_ATTR(hpload_dc_l, S_IRUGO, cs43130_show_dc_l, NULL);
+static DEVICE_ATTR(hpload_dc_r, S_IRUGO, cs43130_show_dc_r, NULL);
+static DEVICE_ATTR(hpload_ac_l, S_IRUGO, cs43130_show_ac_l, NULL);
+static DEVICE_ATTR(hpload_ac_r, S_IRUGO, cs43130_show_ac_r, NULL);
+
+static struct reg_sequence hp_en_cal_seq[] = {
+	{CS43130_INT_MASK_4, CS43130_INT_MASK_ALL},
+	{CS43130_HP_MEAS_LOAD_1, 0},
+	{CS43130_HP_MEAS_LOAD_2, 0},
+	{CS43130_INT_MASK_4, 0},
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD16, 0xBB},
+	{CS43130_DXD12, 0x01},
+	{CS43130_DXD19, 0xCB},
+	{CS43130_DXD17, 0x95},
+	{CS43130_DXD18, 0x0B},
+	{CS43130_DXD1, 0},
+	{CS43130_HP_LOAD_1, 0x80},
+};
+
+static struct reg_sequence hp_en_cal_seq2[] = {
+	{CS43130_INT_MASK_4, CS43130_INT_MASK_ALL},
+	{CS43130_HP_MEAS_LOAD_1, 0},
+	{CS43130_HP_MEAS_LOAD_2, 0},
+	{CS43130_INT_MASK_4, 0},
+	{CS43130_HP_LOAD_1, 0x80},
+};
+
+static struct reg_sequence hp_dis_cal_seq[] = {
+	{CS43130_HP_LOAD_1, 0x80},
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD12, 0},
+	{CS43130_DXD1, 0},
+	{CS43130_HP_LOAD_1, 0},
+};
+
+static struct reg_sequence hp_dis_cal_seq2[] = {
+	{CS43130_HP_LOAD_1, 0x80},
+	{CS43130_HP_LOAD_1, 0},
+};
+
+static struct reg_sequence hp_dc_ch_l_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD19, 0x0A},
+	{CS43130_DXD17, 0x93},
+	{CS43130_DXD18, 0x0A},
+	{CS43130_DXD1, 0},
+	{CS43130_HP_LOAD_1, 0x80},
+	{CS43130_HP_LOAD_1, 0x81},
+};
+
+static struct reg_sequence hp_dc_ch_l_seq2[] = {
+	{CS43130_HP_LOAD_1, 0x80},
+	{CS43130_HP_LOAD_1, 0x81},
+};
+
+static struct reg_sequence hp_dc_ch_r_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD19, 0x8A},
+	{CS43130_DXD17, 0x15},
+	{CS43130_DXD18, 0x06},
+	{CS43130_DXD1, 0},
+	{CS43130_HP_LOAD_1, 0x90},
+	{CS43130_HP_LOAD_1, 0x91},
+};
+
+static struct reg_sequence hp_dc_ch_r_seq2[] = {
+	{CS43130_HP_LOAD_1, 0x90},
+	{CS43130_HP_LOAD_1, 0x91},
+};
+
+static struct reg_sequence hp_ac_ch_l_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD19, 0x0A},
+	{CS43130_DXD17, 0x93},
+	{CS43130_DXD18, 0x0A},
+	{CS43130_DXD1, 0},
+	{CS43130_HP_LOAD_1, 0x80},
+	{CS43130_HP_LOAD_1, 0x82},
+};
+
+static struct reg_sequence hp_ac_ch_l_seq2[] = {
+	{CS43130_HP_LOAD_1, 0x80},
+	{CS43130_HP_LOAD_1, 0x82},
+};
+
+static struct reg_sequence hp_ac_ch_r_seq[] = {
+	{CS43130_DXD1, 0x99},
+	{CS43130_DXD19, 0x8A},
+	{CS43130_DXD17, 0x15},
+	{CS43130_DXD18, 0x06},
+	{CS43130_DXD1, 0},
+	{CS43130_HP_LOAD_1, 0x90},
+	{CS43130_HP_LOAD_1, 0x92},
+};
+
+static struct reg_sequence hp_ac_ch_r_seq2[] = {
+	{CS43130_HP_LOAD_1, 0x90},
+	{CS43130_HP_LOAD_1, 0x92},
+};
+
+static struct reg_sequence hp_cln_seq[] = {
+	{CS43130_INT_MASK_4, CS43130_INT_MASK_ALL},
+	{CS43130_HP_MEAS_LOAD_1, 0},
+	{CS43130_HP_MEAS_LOAD_2, 0},
+};
+
+struct reg_sequences {
+	struct reg_sequence	*seq;
+	int			size;
+	unsigned int		msk;
+};
+
+static struct reg_sequences hpload_seq1[] = {
+	{
+		.seq	= hp_en_cal_seq,
+		.size	= ARRAY_SIZE(hp_en_cal_seq),
+		.msk	= CS43130_HPLOAD_ON_INT,
+	},
+	{
+		.seq	= hp_dc_ch_l_seq,
+		.size	= ARRAY_SIZE(hp_dc_ch_l_seq),
+		.msk	= CS43130_HPLOAD_DC_INT,
+	},
+	{
+		.seq	= hp_ac_ch_l_seq,
+		.size	= ARRAY_SIZE(hp_ac_ch_l_seq),
+		.msk	= CS43130_HPLOAD_AC_INT,
+	},
+	{
+		.seq	= hp_dis_cal_seq,
+		.size	= ARRAY_SIZE(hp_dis_cal_seq),
+		.msk	= CS43130_HPLOAD_OFF_INT,
+	},
+	{
+		.seq	= hp_en_cal_seq,
+		.size	= ARRAY_SIZE(hp_en_cal_seq),
+		.msk	= CS43130_HPLOAD_ON_INT,
+	},
+	{
+		.seq	= hp_dc_ch_r_seq,
+		.size	= ARRAY_SIZE(hp_dc_ch_r_seq),
+		.msk	= CS43130_HPLOAD_DC_INT,
+	},
+	{
+		.seq	= hp_ac_ch_r_seq,
+		.size	= ARRAY_SIZE(hp_ac_ch_r_seq),
+		.msk	= CS43130_HPLOAD_AC_INT,
+	},
+};
+
+static struct reg_sequences hpload_seq2[] = {
+	{
+		.seq	= hp_en_cal_seq2,
+		.size	= ARRAY_SIZE(hp_en_cal_seq2),
+		.msk	= CS43130_HPLOAD_ON_INT,
+	},
+	{
+		.seq	= hp_dc_ch_l_seq2,
+		.size	= ARRAY_SIZE(hp_dc_ch_l_seq2),
+		.msk	= CS43130_HPLOAD_DC_INT,
+	},
+	{
+		.seq	= hp_ac_ch_l_seq2,
+		.size	= ARRAY_SIZE(hp_ac_ch_l_seq2),
+		.msk	= CS43130_HPLOAD_AC_INT,
+	},
+	{
+		.seq	= hp_dis_cal_seq2,
+		.size	= ARRAY_SIZE(hp_dis_cal_seq2),
+		.msk	= CS43130_HPLOAD_OFF_INT,
+	},
+	{
+		.seq	= hp_en_cal_seq2,
+		.size	= ARRAY_SIZE(hp_en_cal_seq2),
+		.msk	= CS43130_HPLOAD_ON_INT,
+	},
+	{
+		.seq	= hp_dc_ch_r_seq2,
+		.size	= ARRAY_SIZE(hp_dc_ch_r_seq2),
+		.msk	= CS43130_HPLOAD_DC_INT,
+	},
+	{
+		.seq	= hp_ac_ch_r_seq2,
+		.size	= ARRAY_SIZE(hp_ac_ch_r_seq2),
+		.msk	= CS43130_HPLOAD_AC_INT,
+	},
+};
+
+static int cs43130_update_hpload(unsigned int msk, int ac_idx,
+				 struct cs43130_private *cs43130)
+{
+	bool left_ch = true;
+	unsigned int reg;
+	u32 addr;
+	u16 impedance;
+	struct snd_soc_codec *codec = cs43130->codec;
+
+	switch (msk) {
+	case CS43130_HPLOAD_DC_INT:
+	case CS43130_HPLOAD_AC_INT:
+		break;
+	default:
+		return 0;
+	}
+
+	regmap_read(cs43130->regmap, CS43130_HP_LOAD_1, &reg);
+	if (reg & CS43130_HPLOAD_CHN_SEL)
+		left_ch = false;
+
+	if (msk == CS43130_HPLOAD_DC_INT)
+		addr = CS43130_HP_DC_STAT_1;
+	else
+		addr = CS43130_HP_AC_STAT_1;
+
+	regmap_read(cs43130->regmap, addr, &reg);
+	impedance = reg >> 3;
+	regmap_read(cs43130->regmap, addr + 1, &reg);
+	impedance |= reg << 5;
+
+	if (msk == CS43130_HPLOAD_DC_INT) {
+		if (left_ch)
+			cs43130->hpload_dc[HP_LEFT] = impedance;
+		else
+			cs43130->hpload_dc[HP_RIGHT] = impedance;
+
+		dev_dbg(codec->dev, "HP DC impedance (Ch %u): %u\n", !left_ch,
+			impedance);
+	} else {
+		if (left_ch)
+			cs43130->hpload_ac[ac_idx][HP_LEFT] = impedance;
+		else
+			cs43130->hpload_ac[ac_idx][HP_RIGHT] = impedance;
+
+		dev_dbg(codec->dev, "HP AC (%u Hz) impedance (Ch %u): %u\n",
+			cs43130->ac_freq[ac_idx], !left_ch, impedance);
+	}
+
+	return 0;
+}
+
+static int cs43130_hpload_proc(struct cs43130_private *cs43130,
+			       struct reg_sequence *seq, int seq_size,
+			       unsigned int rslt_msk, int ac_idx)
+{
+	int ret;
+	unsigned int msk;
+	u16 ac_reg_val;
+	struct snd_soc_codec *codec = cs43130->codec;
+
+	reinit_completion(&cs43130->hpload_evt);
+
+	if (rslt_msk == CS43130_HPLOAD_AC_INT) {
+		ac_reg_val = cs43130_get_ac_reg_val(cs43130->ac_freq[ac_idx]);
+		regmap_update_bits(cs43130->regmap, CS43130_HP_LOAD_1,
+				   CS43130_HPLOAD_AC_START, 0);
+		regmap_update_bits(cs43130->regmap, CS43130_HP_MEAS_LOAD_1,
+				   CS43130_HP_MEAS_LOAD_MASK,
+				   ac_reg_val >> CS43130_HP_MEAS_LOAD_1_SHIFT);
+		regmap_update_bits(cs43130->regmap, CS43130_HP_MEAS_LOAD_2,
+				   CS43130_HP_MEAS_LOAD_MASK,
+				   ac_reg_val >> CS43130_HP_MEAS_LOAD_2_SHIFT);
+	}
+
+	regmap_multi_reg_write(cs43130->regmap, seq,
+			       seq_size);
+
+	ret = wait_for_completion_timeout(&cs43130->hpload_evt,
+					  msecs_to_jiffies(1000));
+	regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk);
+	if (!ret) {
+		dev_err(codec->dev, "Timeout waiting for HPLOAD interrupt\n");
+		return -1;
+	}
+
+	dev_dbg(codec->dev, "HP load stat: %x, INT_MASK_4: %x\n",
+		cs43130->hpload_stat, msk);
+	if ((cs43130->hpload_stat & (CS43130_HPLOAD_NO_DC_INT |
+				     CS43130_HPLOAD_UNPLUG_INT |
+				     CS43130_HPLOAD_OOR_INT)) ||
+	    !(cs43130->hpload_stat & rslt_msk)) {
+		dev_dbg(codec->dev, "HP load measure failed\n");
+		return -1;
+	}
+
+	return 0;
+}
+
+static const struct reg_sequence hv_seq[][2] = {
+	{
+		{CS43130_CLASS_H_CTL, 0x1C},
+		{CS43130_HP_OUT_CTL_1, 0x10},
+	},
+	{
+		{CS43130_CLASS_H_CTL, 0x1E},
+		{CS43130_HP_OUT_CTL_1, 0x20},
+	},
+	{
+		{CS43130_CLASS_H_CTL, 0x1E},
+		{CS43130_HP_OUT_CTL_1, 0x30},
+	},
+};
+
+static int cs43130_set_hv(struct regmap *regmap, u16 hpload_dc,
+			  const u16 *dc_threshold)
+{
+	int i;
+
+	for (i = 0; i < CS43130_DC_THRESHOLD; i++) {
+		if (hpload_dc <= dc_threshold[i])
+			break;
+	}
+
+	regmap_multi_reg_write(regmap, hv_seq[i], ARRAY_SIZE(hv_seq[i]));
+
+	return 0;
+}
+
+static void cs43130_imp_meas(struct work_struct *wk)
+{
+	unsigned int reg, seq_size;
+	int i, ret, ac_idx;
+	struct cs43130_private *cs43130;
+	struct snd_soc_codec *codec;
+	struct reg_sequences *hpload_seq;
+
+	cs43130 = container_of(wk, struct cs43130_private, work);
+	codec = cs43130->codec;
+
+	if (!cs43130->mclk)
+		return;
+
+	cs43130->hpload_done = false;
+
+	mutex_lock(&cs43130->clk_mutex);
+	if (!cs43130->clk_req) {
+		/* clk not in use */
+		cs43130_set_pll(codec, 0, 0, cs43130->mclk, CS43130_MCLK_22M);
+		if (cs43130->pll_bypass)
+			cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT);
+		else
+			cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL);
+	}
+
+	cs43130->clk_req++;
+	mutex_unlock(&cs43130->clk_mutex);
+
+	regmap_read(cs43130->regmap, CS43130_INT_STATUS_4, &reg);
+
+	switch (cs43130->dev_id) {
+	case CS43130_CHIP_ID:
+		hpload_seq = hpload_seq1;
+		seq_size = ARRAY_SIZE(hpload_seq1);
+		break;
+	case CS43131_CHIP_ID:
+		hpload_seq = hpload_seq2;
+		seq_size = ARRAY_SIZE(hpload_seq2);
+	}
+
+	i = 0;
+	ac_idx = 0;
+	while (i < seq_size) {
+		ret = cs43130_hpload_proc(cs43130, hpload_seq[i].seq,
+					  hpload_seq[i].size,
+					  hpload_seq[i].msk, ac_idx);
+		if (ret < 0)
+			goto exit;
+
+		cs43130_update_hpload(hpload_seq[i].msk, ac_idx, cs43130);
+
+		if (cs43130->ac_meas &&
+		    hpload_seq[i].msk == CS43130_HPLOAD_AC_INT &&
+		    ac_idx < CS43130_AC_FREQ - 1) {
+			ac_idx++;
+		} else {
+			ac_idx = 0;
+			i++;
+		}
+	}
+	cs43130->hpload_done = true;
+
+	if (cs43130->hpload_dc[HP_LEFT] >= CS43130_LINEOUT_LOAD)
+		snd_soc_jack_report(&cs43130->jack, CS43130_JACK_LINEOUT,
+				    CS43130_JACK_MASK);
+	else
+		snd_soc_jack_report(&cs43130->jack, CS43130_JACK_HEADPHONE,
+				    CS43130_JACK_MASK);
+
+	dev_dbg(codec->dev, "Set HP output control. DC threshold\n");
+	for (i = 0; i < CS43130_DC_THRESHOLD; i++)
+		dev_dbg(codec->dev, "DC threshold[%d]: %u.\n", i,
+			cs43130->dc_threshold[i]);
+
+	cs43130_set_hv(cs43130->regmap, cs43130->hpload_dc[HP_LEFT],
+		       cs43130->dc_threshold);
+
+exit:
+	switch (cs43130->dev_id) {
+	case CS43130_CHIP_ID:
+		cs43130_hpload_proc(cs43130, hp_dis_cal_seq,
+				    ARRAY_SIZE(hp_dis_cal_seq),
+				    CS43130_HPLOAD_OFF_INT, ac_idx);
+		break;
+	case CS43131_CHIP_ID:
+		cs43130_hpload_proc(cs43130, hp_dis_cal_seq2,
+				    ARRAY_SIZE(hp_dis_cal_seq2),
+				    CS43130_HPLOAD_OFF_INT, ac_idx);
+	}
+
+	regmap_multi_reg_write(cs43130->regmap, hp_cln_seq,
+			       ARRAY_SIZE(hp_cln_seq));
+
+	mutex_lock(&cs43130->clk_mutex);
+	cs43130->clk_req--;
+	/* clk not in use */
+	if (!cs43130->clk_req)
+		cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO);
+	mutex_unlock(&cs43130->clk_mutex);
+}
+
+static irqreturn_t cs43130_irq_thread(int irq, void *data)
+{
+	struct cs43130_private *cs43130 = (struct cs43130_private *)data;
+	struct snd_soc_codec *codec = cs43130->codec;
+	unsigned int stickies[CS43130_NUM_INT];
+	unsigned int irq_occurrance = 0;
+	unsigned int masks[CS43130_NUM_INT];
+	int i, j;
+
+	for (i = 0; i < ARRAY_SIZE(stickies); i++) {
+		regmap_read(cs43130->regmap, CS43130_INT_STATUS_1 + i,
+			    &stickies[i]);
+		regmap_read(cs43130->regmap, CS43130_INT_MASK_1 + i,
+			    &masks[i]);
+	}
+
+	for (i = 0; i < ARRAY_SIZE(stickies); i++) {
+		stickies[i] = stickies[i] & (~masks[i]);
+		for (j = 0; j < 8; j++)
+			irq_occurrance += (stickies[i] >> j) & 1;
+	}
+	dev_dbg(codec->dev, "number of interrupts occurred (%u)\n",
+		irq_occurrance);
+
+	if (!irq_occurrance)
+		return IRQ_NONE;
+
+	if (stickies[0] & CS43130_XTAL_RDY_INT) {
+		complete(&cs43130->xtal_rdy);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[0] & CS43130_PLL_RDY_INT) {
+		complete(&cs43130->pll_rdy);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[3] & CS43130_HPLOAD_NO_DC_INT) {
+		cs43130->hpload_stat = stickies[3];
+		dev_err(codec->dev,
+			"DC load has not completed before AC load (%x)\n",
+			cs43130->hpload_stat);
+		complete(&cs43130->hpload_evt);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[3] & CS43130_HPLOAD_UNPLUG_INT) {
+		cs43130->hpload_stat = stickies[3];
+		dev_err(codec->dev, "HP unplugged during measurement (%x)\n",
+			cs43130->hpload_stat);
+		complete(&cs43130->hpload_evt);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[3] & CS43130_HPLOAD_OOR_INT) {
+		cs43130->hpload_stat = stickies[3];
+		dev_err(codec->dev, "HP load out of range (%x)\n",
+			cs43130->hpload_stat);
+		complete(&cs43130->hpload_evt);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[3] & CS43130_HPLOAD_AC_INT) {
+		cs43130->hpload_stat = stickies[3];
+		dev_dbg(codec->dev, "HP AC load measurement done (%x)\n",
+			cs43130->hpload_stat);
+		complete(&cs43130->hpload_evt);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[3] & CS43130_HPLOAD_DC_INT) {
+		cs43130->hpload_stat = stickies[3];
+		dev_dbg(codec->dev, "HP DC load measurement done (%x)\n",
+			cs43130->hpload_stat);
+		complete(&cs43130->hpload_evt);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[3] & CS43130_HPLOAD_ON_INT) {
+		cs43130->hpload_stat = stickies[3];
+		dev_dbg(codec->dev, "HP load state machine on done (%x)\n",
+			cs43130->hpload_stat);
+		complete(&cs43130->hpload_evt);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[3] & CS43130_HPLOAD_OFF_INT) {
+		cs43130->hpload_stat = stickies[3];
+		dev_dbg(codec->dev, "HP load state machine off done (%x)\n",
+			cs43130->hpload_stat);
+		complete(&cs43130->hpload_evt);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[0] & CS43130_XTAL_ERR_INT) {
+		dev_err(codec->dev, "Crystal err: clock is not running\n");
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[0] & CS43130_HP_UNPLUG_INT) {
+		dev_dbg(codec->dev, "HP unplugged\n");
+		cs43130->hpload_done = false;
+		snd_soc_jack_report(&cs43130->jack, 0, CS43130_JACK_MASK);
+		return IRQ_HANDLED;
+	}
+
+	if (stickies[0] & CS43130_HP_PLUG_INT) {
+		if (cs43130->dc_meas && !cs43130->hpload_done &&
+		    !work_busy(&cs43130->work)) {
+			dev_dbg(codec->dev, "HP load queue work\n");
+			queue_work(cs43130->wq, &cs43130->work);
+		}
+
+		snd_soc_jack_report(&cs43130->jack, SND_JACK_MECHANICAL,
+				    CS43130_JACK_MASK);
+		return IRQ_HANDLED;
+	}
+
+	return IRQ_NONE;
+}
+
+static int cs43130_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_card *card = codec->component.card;
+	unsigned int reg;
+
+	cs43130->codec = codec;
+
+	if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED) {
+		regmap_update_bits(cs43130->regmap, CS43130_CRYSTAL_SET,
+				   CS43130_XTAL_IBIAS_MASK,
+				   cs43130->xtal_ibias);
+		regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+				   CS43130_XTAL_ERR_INT, 0);
+	}
+
+	ret = snd_soc_card_jack_new(card, "Headphone", CS43130_JACK_MASK,
+				    &cs43130->jack, NULL, 0);
+	if (ret < 0) {
+		dev_err(codec->dev, "Cannot create jack\n");
+		return ret;
+	}
+
+	cs43130->hpload_done = false;
+	if (cs43130->dc_meas) {
+		ret = device_create_file(codec->dev, &dev_attr_hpload_dc_l);
+		if (ret < 0)
+			return ret;
+
+		ret = device_create_file(codec->dev, &dev_attr_hpload_dc_r);
+		if (ret < 0)
+			return ret;
+
+		ret = device_create_file(codec->dev, &dev_attr_hpload_ac_l);
+		if (ret < 0)
+			return ret;
+
+		ret = device_create_file(codec->dev, &dev_attr_hpload_ac_r);
+		if (ret < 0)
+			return ret;
+
+		cs43130->wq = create_singlethread_workqueue("cs43130_hp");
+		INIT_WORK(&cs43130->work, cs43130_imp_meas);
+	}
+
+	regmap_read(cs43130->regmap, CS43130_INT_STATUS_1, &reg);
+	regmap_read(cs43130->regmap, CS43130_HP_STATUS, &reg);
+	regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+			   CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT, 0);
+	regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT,
+			   CS43130_HP_DETECT_CTRL_MASK, 0);
+	regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT,
+			   CS43130_HP_DETECT_CTRL_MASK,
+			   CS43130_HP_DETECT_CTRL_MASK);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs43130 = {
+	.probe			= cs43130_probe,
+	.component_driver = {
+		.controls		= cs43130_snd_controls,
+		.num_controls		= ARRAY_SIZE(cs43130_snd_controls),
+	},
+	.set_sysclk		= cs43130_codec_set_sysclk,
+	.set_pll		= cs43130_set_pll,
+};
+
+static const struct regmap_config cs43130_regmap = {
+	.reg_bits		= 24,
+	.pad_bits		= 8,
+	.val_bits		= 8,
+
+	.max_register		= CS43130_LASTREG,
+	.reg_defaults		= cs43130_reg_defaults,
+	.num_reg_defaults	= ARRAY_SIZE(cs43130_reg_defaults),
+	.readable_reg		= cs43130_readable_register,
+	.precious_reg		= cs43130_precious_register,
+	.volatile_reg		= cs43130_volatile_register,
+	.cache_type		= REGCACHE_RBTREE,
+	.use_single_rw		= true, /* needed for regcache_sync */
+};
+
+static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = {
+	50,
+	120,
+};
+
+static int cs43130_handle_device_data(struct i2c_client *i2c_client,
+				      struct cs43130_private *cs43130)
+{
+	struct device_node *np = i2c_client->dev.of_node;
+	unsigned int val;
+	int i;
+
+	if (of_property_read_u32(np, "cirrus,xtal-ibias", &val) < 0) {
+		/* Crystal is unused. System clock is used for external MCLK */
+		cs43130->xtal_ibias = CS43130_XTAL_UNUSED;
+		return 0;
+	}
+
+	switch (val) {
+	case 1:
+		cs43130->xtal_ibias = CS43130_XTAL_IBIAS_7_5UA;
+		break;
+	case 2:
+		cs43130->xtal_ibias = CS43130_XTAL_IBIAS_12_5UA;
+		break;
+	case 3:
+		cs43130->xtal_ibias = CS43130_XTAL_IBIAS_15UA;
+		break;
+	default:
+		dev_err(&i2c_client->dev,
+			"Invalid cirrus,xtal-ibias value: %d\n", val);
+		return -EINVAL;
+	}
+
+	cs43130->dc_meas = of_property_read_bool(np, "cirrus,dc-measure");
+	cs43130->ac_meas = of_property_read_bool(np, "cirrus,ac-measure");
+
+	if (of_property_read_u16_array(np, "cirrus,ac-freq", cs43130->ac_freq,
+					CS43130_AC_FREQ) < 0) {
+		for (i = 0; i < CS43130_AC_FREQ; i++)
+			cs43130->ac_freq[i] = cs43130_ac_freq[i];
+	}
+
+	if (of_property_read_u16_array(np, "cirrus,dc-threshold",
+				       cs43130->dc_threshold,
+				       CS43130_DC_THRESHOLD) < 0) {
+		for (i = 0; i < CS43130_DC_THRESHOLD; i++)
+			cs43130->dc_threshold[i] = cs43130_dc_threshold[i];
+	}
+
+	return 0;
+}
+
+static int cs43130_i2c_probe(struct i2c_client *client,
+			     const struct i2c_device_id *id)
+{
+	struct cs43130_private *cs43130;
+	int ret;
+	unsigned int devid = 0;
+	unsigned int reg;
+	int i;
+
+	cs43130 = devm_kzalloc(&client->dev, sizeof(*cs43130), GFP_KERNEL);
+	if (!cs43130)
+		return -ENOMEM;
+
+	i2c_set_clientdata(client, cs43130);
+
+	cs43130->regmap = devm_regmap_init_i2c(client, &cs43130_regmap);
+	if (IS_ERR(cs43130->regmap)) {
+		ret = PTR_ERR(cs43130->regmap);
+		return ret;
+	}
+
+	if (client->dev.of_node) {
+		ret = cs43130_handle_device_data(client, cs43130);
+		if (ret != 0)
+			return ret;
+	}
+	for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++)
+		cs43130->supplies[i].supply = cs43130_supply_names[i];
+
+	ret = devm_regulator_bulk_get(&client->dev,
+				      ARRAY_SIZE(cs43130->supplies),
+				      cs43130->supplies);
+	if (ret != 0) {
+		dev_err(&client->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+	ret = regulator_bulk_enable(ARRAY_SIZE(cs43130->supplies),
+				    cs43130->supplies);
+	if (ret != 0) {
+		dev_err(&client->dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	cs43130->reset_gpio = devm_gpiod_get_optional(&client->dev,
+						      "reset", GPIOD_OUT_LOW);
+	if (IS_ERR(cs43130->reset_gpio))
+		return PTR_ERR(cs43130->reset_gpio);
+
+	gpiod_set_value_cansleep(cs43130->reset_gpio, 1);
+
+	usleep_range(2000, 2050);
+
+	ret = regmap_read(cs43130->regmap, CS43130_DEVID_AB, &reg);
+
+	devid = (reg & 0xFF) << 12;
+	ret = regmap_read(cs43130->regmap, CS43130_DEVID_CD, &reg);
+	devid |= (reg & 0xFF) << 4;
+	ret = regmap_read(cs43130->regmap, CS43130_DEVID_E, &reg);
+	devid |= (reg & 0xF0) >> 4;
+
+	switch (devid) {
+	case CS43130_CHIP_ID:
+	case CS4399_CHIP_ID:
+	case CS43131_CHIP_ID:
+	case CS43198_CHIP_ID:
+		break;
+	default:
+		dev_err(&client->dev,
+			"CS43130 Device ID %X. Expected ID %X, %X, %X or %X\n",
+			devid, CS43130_CHIP_ID, CS4399_CHIP_ID,
+			CS43131_CHIP_ID, CS43198_CHIP_ID);
+		ret = -ENODEV;
+		goto err;
+	}
+
+	cs43130->dev_id = devid;
+	ret = regmap_read(cs43130->regmap, CS43130_REV_ID, &reg);
+	if (ret < 0) {
+		dev_err(&client->dev, "Get Revision ID failed\n");
+		goto err;
+	}
+
+	dev_info(&client->dev,
+		 "Cirrus Logic CS43130 (%x), Revision: %02X\n", devid,
+		 reg & 0xFF);
+
+	mutex_init(&cs43130->clk_mutex);
+
+	init_completion(&cs43130->xtal_rdy);
+	init_completion(&cs43130->pll_rdy);
+	init_completion(&cs43130->hpload_evt);
+
+	ret = devm_request_threaded_irq(&client->dev, client->irq,
+					NULL, cs43130_irq_thread,
+					IRQF_ONESHOT | IRQF_TRIGGER_LOW,
+					"cs43130", cs43130);
+	if (ret != 0) {
+		dev_err(&client->dev, "Failed to request IRQ: %d\n", ret);
+		return ret;
+	}
+
+	cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO;
+
+	pm_runtime_set_autosuspend_delay(&client->dev, 100);
+	pm_runtime_use_autosuspend(&client->dev);
+	pm_runtime_set_active(&client->dev);
+	pm_runtime_enable(&client->dev);
+
+	switch (cs43130->dev_id) {
+	case CS43130_CHIP_ID:
+	case CS43131_CHIP_ID:
+		memcpy(all_hp_widgets, digital_hp_widgets,
+		       sizeof(digital_hp_widgets));
+		memcpy(all_hp_widgets + ARRAY_SIZE(digital_hp_widgets),
+		       analog_hp_widgets, sizeof(analog_hp_widgets));
+		memcpy(all_hp_routes, digital_hp_routes,
+		       sizeof(digital_hp_routes));
+		memcpy(all_hp_routes + ARRAY_SIZE(digital_hp_routes),
+		       analog_hp_routes, sizeof(analog_hp_routes));
+
+		soc_codec_dev_cs43130.component_driver.dapm_widgets =
+			all_hp_widgets;
+		soc_codec_dev_cs43130.component_driver.num_dapm_widgets =
+			ARRAY_SIZE(all_hp_widgets);
+		soc_codec_dev_cs43130.component_driver.dapm_routes =
+			all_hp_routes;
+		soc_codec_dev_cs43130.component_driver.num_dapm_routes =
+			ARRAY_SIZE(all_hp_routes);
+		break;
+	case CS43198_CHIP_ID:
+	case CS4399_CHIP_ID:
+		soc_codec_dev_cs43130.component_driver.dapm_widgets =
+			digital_hp_widgets;
+		soc_codec_dev_cs43130.component_driver.num_dapm_widgets =
+			ARRAY_SIZE(digital_hp_widgets);
+		soc_codec_dev_cs43130.component_driver.dapm_routes =
+			digital_hp_routes;
+		soc_codec_dev_cs43130.component_driver.num_dapm_routes =
+			ARRAY_SIZE(digital_hp_routes);
+	}
+
+	ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_cs43130,
+				     cs43130_dai, ARRAY_SIZE(cs43130_dai));
+	if (ret < 0) {
+		dev_err(&client->dev,
+			"snd_soc_register_codec failed with ret = %d\n", ret);
+		goto err;
+	}
+
+	regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG,
+			   CS43130_ASP_3ST_MASK, 0);
+	regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG,
+			   CS43130_XSP_3ST_MASK, 0);
+
+	return 0;
+err:
+	return ret;
+}
+
+static int cs43130_i2c_remove(struct i2c_client *client)
+{
+	struct cs43130_private *cs43130 = i2c_get_clientdata(client);
+
+	if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED)
+		regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+				   CS43130_XTAL_ERR_INT,
+				   1 << CS43130_XTAL_ERR_INT_SHIFT);
+
+	regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+			   CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT,
+			   CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT);
+
+	if (cs43130->dc_meas) {
+		cancel_work_sync(&cs43130->work);
+		flush_workqueue(cs43130->wq);
+
+		device_remove_file(&client->dev, &dev_attr_hpload_dc_l);
+		device_remove_file(&client->dev, &dev_attr_hpload_dc_r);
+		device_remove_file(&client->dev, &dev_attr_hpload_ac_l);
+		device_remove_file(&client->dev, &dev_attr_hpload_ac_r);
+	}
+
+	if (cs43130->reset_gpio)
+		gpiod_set_value_cansleep(cs43130->reset_gpio, 0);
+
+	pm_runtime_disable(&client->dev);
+	regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies);
+
+	snd_soc_unregister_codec(&client->dev);
+
+	return 0;
+}
+
+static int cs43130_runtime_suspend(struct device *dev)
+{
+	struct cs43130_private *cs43130 = dev_get_drvdata(dev);
+
+	if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED)
+		regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+				   CS43130_XTAL_ERR_INT,
+				   1 << CS43130_XTAL_ERR_INT_SHIFT);
+
+	regcache_cache_only(cs43130->regmap, true);
+	regcache_mark_dirty(cs43130->regmap);
+
+	gpiod_set_value_cansleep(cs43130->reset_gpio, 0);
+
+	regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies);
+
+	return 0;
+}
+
+static int cs43130_runtime_resume(struct device *dev)
+{
+	struct cs43130_private *cs43130 = dev_get_drvdata(dev);
+	int ret;
+
+	ret = regulator_bulk_enable(CS43130_NUM_SUPPLIES, cs43130->supplies);
+	if (ret != 0) {
+		dev_err(dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	regcache_cache_only(cs43130->regmap, false);
+
+	gpiod_set_value_cansleep(cs43130->reset_gpio, 1);
+
+	usleep_range(2000, 2050);
+
+	ret = regcache_sync(cs43130->regmap);
+	if (ret != 0) {
+		dev_err(dev, "Failed to restore register cache\n");
+		goto err;
+	}
+
+	if (cs43130->xtal_ibias != CS43130_XTAL_UNUSED)
+		regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1,
+				   CS43130_XTAL_ERR_INT, 0);
+
+	return 0;
+err:
+	regcache_cache_only(cs43130->regmap, true);
+	regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies);
+
+	return ret;
+}
+
+static const struct dev_pm_ops cs43130_runtime_pm = {
+	SET_RUNTIME_PM_OPS(cs43130_runtime_suspend, cs43130_runtime_resume,
+			   NULL)
+};
+
+static const struct of_device_id cs43130_of_match[] = {
+	{.compatible = "cirrus,cs43130",},
+	{.compatible = "cirrus,cs4399",},
+	{.compatible = "cirrus,cs43131",},
+	{.compatible = "cirrus,cs43198",},
+	{},
+};
+
+MODULE_DEVICE_TABLE(of, cs43130_of_match);
+
+static const struct i2c_device_id cs43130_i2c_id[] = {
+	{"cs43130", 0},
+	{"cs4399", 0},
+	{"cs43131", 0},
+	{"cs43198", 0},
+	{}
+};
+
+MODULE_DEVICE_TABLE(i2c, cs43130_i2c_id);
+
+static struct i2c_driver cs43130_i2c_driver = {
+	.driver = {
+		.name		= "cs43130",
+		.of_match_table	= cs43130_of_match,
+		.pm             = &cs43130_runtime_pm,
+	},
+	.id_table	= cs43130_i2c_id,
+	.probe		= cs43130_i2c_probe,
+	.remove		= cs43130_i2c_remove,
+};
+
+module_i2c_driver(cs43130_i2c_driver);
+
+MODULE_AUTHOR("Li Xu <li.xu@cirrus.com>");
+MODULE_DESCRIPTION("Cirrus Logic CS43130 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h
new file mode 100644
index 000000000000..781258418d89
--- /dev/null
+++ b/sound/soc/codecs/cs43130.h
@@ -0,0 +1,546 @@
+/*
+ * ALSA SoC CS43130 codec driver
+ *
+ * Copyright 2017 Cirrus Logic, Inc.
+ *
+ * Author: Li Xu <li.xu@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ */
+
+#ifndef __CS43130_H__
+#define __CS43130_H__
+
+/* CS43130 registers addresses */
+/* all reg address is shifted by a byte for control byte to be LSB */
+#define CS43130_FIRSTREG	0x010000
+#define CS43130_LASTREG		0x190000
+#define CS43130_CHIP_ID		0x00043130
+#define CS4399_CHIP_ID		0x00043990
+#define CS43131_CHIP_ID		0x00043131
+#define CS43198_CHIP_ID		0x00043198
+#define CS43130_DEVID_AB	0x010000	/* Device ID A & B [RO] */
+#define CS43130_DEVID_CD	0x010001	/* Device ID C & D [RO] */
+#define CS43130_DEVID_E		0x010002	/* Device ID E [RO] */
+#define CS43130_FAB_ID		0x010003        /* Fab ID [RO] */
+#define CS43130_REV_ID		0x010004        /* Revision ID [RO] */
+#define CS43130_SUBREV_ID	0x010005        /* Subrevision ID */
+#define CS43130_SYS_CLK_CTL_1	0x010006	/* System Clocking Ctl 1 */
+#define CS43130_SP_SRATE	0x01000B        /* Serial Port Sample Rate */
+#define CS43130_SP_BITSIZE	0x01000C        /* Serial Port Bit Size */
+#define CS43130_PAD_INT_CFG	0x01000D	/* Pad Interface Config */
+#define CS43130_DXD1            0x010010        /* DXD1 */
+#define CS43130_DXD7            0x010025        /* DXD7 */
+#define CS43130_DXD19           0x010026        /* DXD19 */
+#define CS43130_DXD17           0x010027        /* DXD17 */
+#define CS43130_DXD18           0x010028        /* DXD18 */
+#define CS43130_DXD12           0x01002C        /* DXD12 */
+#define CS43130_DXD8            0x01002E        /* DXD8 */
+#define CS43130_PWDN_CTL	0x020000        /* Power Down Ctl */
+#define CS43130_DXD2            0x020019        /* DXD2 */
+#define CS43130_CRYSTAL_SET	0x020052	/* Crystal Setting */
+#define CS43130_PLL_SET_1	0x030001        /* PLL Setting 1 */
+#define CS43130_PLL_SET_2	0x030002        /* PLL Setting 2 */
+#define CS43130_PLL_SET_3	0x030003        /* PLL Setting 3 */
+#define CS43130_PLL_SET_4	0x030004        /* PLL Setting 4 */
+#define CS43130_PLL_SET_5	0x030005        /* PLL Setting 5 */
+#define CS43130_PLL_SET_6	0x030008        /* PLL Setting 6 */
+#define CS43130_PLL_SET_7	0x03000A        /* PLL Setting 7 */
+#define CS43130_PLL_SET_8	0x03001B        /* PLL Setting 8 */
+#define CS43130_PLL_SET_9	0x040002        /* PLL Setting 9 */
+#define CS43130_PLL_SET_10	0x040003        /* PLL Setting 10 */
+#define CS43130_CLKOUT_CTL	0x040004        /* CLKOUT Ctl */
+#define CS43130_ASP_NUM_1	0x040010        /* ASP Numerator 1 */
+#define CS43130_ASP_NUM_2	0x040011        /* ASP Numerator 2 */
+#define CS43130_ASP_DEN_1	0x040012	/* ASP Denominator 1 */
+#define CS43130_ASP_DEN_2	0x040013	/* ASP Denominator 2 */
+#define CS43130_ASP_LRCK_HI_TIME_1 0x040014	/* ASP LRCK High Time 1 */
+#define CS43130_ASP_LRCK_HI_TIME_2 0x040015	/* ASP LRCK High Time 2 */
+#define CS43130_ASP_LRCK_PERIOD_1  0x040016	/* ASP LRCK Period 1 */
+#define CS43130_ASP_LRCK_PERIOD_2  0x040017	/* ASP LRCK Period 2 */
+#define CS43130_ASP_CLOCK_CONF	0x040018	/* ASP Clock Config */
+#define CS43130_ASP_FRAME_CONF	0x040019	/* ASP Frame Config */
+#define CS43130_XSP_NUM_1	0x040020        /* XSP Numerator 1 */
+#define CS43130_XSP_NUM_2	0x040021        /* XSP Numerator 2 */
+#define CS43130_XSP_DEN_1	0x040022	/* XSP Denominator 1 */
+#define CS43130_XSP_DEN_2	0x040023	/* XSP Denominator 2 */
+#define CS43130_XSP_LRCK_HI_TIME_1 0x040024	/* XSP LRCK High Time 1 */
+#define CS43130_XSP_LRCK_HI_TIME_2 0x040025	/* XSP LRCK High Time 2 */
+#define CS43130_XSP_LRCK_PERIOD_1  0x040026	/* XSP LRCK Period 1 */
+#define CS43130_XSP_LRCK_PERIOD_2  0x040027	/* XSP LRCK Period 2 */
+#define CS43130_XSP_CLOCK_CONF	0x040028	/* XSP Clock Config */
+#define CS43130_XSP_FRAME_CONF	0x040029	/* XSP Frame Config */
+#define CS43130_ASP_CH_1_LOC	0x050000	/* ASP Chan 1 Location */
+#define CS43130_ASP_CH_2_LOC	0x050001	/* ASP Chan 2 Location */
+#define CS43130_ASP_CH_1_SZ_EN	0x05000A	/* ASP Chan 1 Size, Enable */
+#define CS43130_ASP_CH_2_SZ_EN	0x05000B	/* ASP Chan 2 Size, Enable */
+#define CS43130_XSP_CH_1_LOC	0x060000	/* XSP Chan 1 Location */
+#define CS43130_XSP_CH_2_LOC	0x060001	/* XSP Chan 2 Location */
+#define CS43130_XSP_CH_1_SZ_EN	0x06000A	/* XSP Chan 1 Size, Enable */
+#define CS43130_XSP_CH_2_SZ_EN	0x06000B	/* XSP Chan 2 Size, Enable */
+#define CS43130_DSD_VOL_B	0x070000        /* DSD Volume B */
+#define CS43130_DSD_VOL_A	0x070001        /* DSD Volume A */
+#define CS43130_DSD_PATH_CTL_1	0x070002	/* DSD Proc Path Sig Ctl 1 */
+#define CS43130_DSD_INT_CFG	0x070003	/* DSD Interface Config */
+#define CS43130_DSD_PATH_CTL_2	0x070004	/* DSD Proc Path Sig Ctl 2 */
+#define CS43130_DSD_PCM_MIX_CTL	0x070005	/* DSD and PCM Mixing Ctl */
+#define CS43130_DSD_PATH_CTL_3	0x070006	/* DSD Proc Path Sig Ctl 3 */
+#define CS43130_HP_OUT_CTL_1	0x080000	/* HP Output Ctl 1 */
+#define CS43130_DXD16		0x080024	/* DXD16 */
+#define CS43130_DXD13		0x080032	/* DXD13 */
+#define CS43130_PCM_FILT_OPT	0x090000	/* PCM Filter Option */
+#define CS43130_PCM_VOL_B	0x090001        /* PCM Volume B */
+#define CS43130_PCM_VOL_A	0x090002        /* PCM Volume A */
+#define CS43130_PCM_PATH_CTL_1	0x090003	/* PCM Path Signal Ctl 1 */
+#define CS43130_PCM_PATH_CTL_2	0x090004	/* PCM Path Signal Ctl 2 */
+#define CS43130_DXD6		0x090097	/* DXD6 */
+#define CS43130_CLASS_H_CTL	0x0B0000	/* Class H Ctl */
+#define CS43130_DXD15		0x0B0005	/* DXD15 */
+#define CS43130_DXD14		0x0B0006	/* DXD14 */
+#define CS43130_DXD3		0x0C0002	/* DXD3 */
+#define CS43130_DXD10		0x0C0003	/* DXD10 */
+#define CS43130_DXD11		0x0C0005	/* DXD11 */
+#define CS43130_DXD9		0x0C0006	/* DXD9 */
+#define CS43130_DXD4		0x0C0009	/* DXD4 */
+#define CS43130_DXD5		0x0C000E	/* DXD5 */
+#define CS43130_HP_DETECT	0x0D0000        /* HP Detect */
+#define CS43130_HP_STATUS	0x0D0001        /* HP Status [RO] */
+#define CS43130_HP_LOAD_1	0x0E0000        /* HP Load 1 */
+#define CS43130_HP_MEAS_LOAD_1	0x0E0003	/* HP Load Measurement 1 */
+#define CS43130_HP_MEAS_LOAD_2	0x0E0004	/* HP Load Measurement 2 */
+#define CS43130_HP_DC_STAT_1	0x0E000D	/* HP DC Load Status 0 [RO] */
+#define CS43130_HP_DC_STAT_2	0x0E000E	/* HP DC Load Status 1 [RO] */
+#define CS43130_HP_AC_STAT_1	0x0E0010	/* HP AC Load Status 0 [RO] */
+#define CS43130_HP_AC_STAT_2	0x0E0011	/* HP AC Load Status 1 [RO] */
+#define CS43130_HP_LOAD_STAT	0x0E001A	/* HP Load Status [RO] */
+#define CS43130_INT_STATUS_1	0x0F0000	/* Interrupt Status 1 */
+#define CS43130_INT_STATUS_2	0x0F0001	/* Interrupt Status 2 */
+#define CS43130_INT_STATUS_3	0x0F0002	/* Interrupt Status 3 */
+#define CS43130_INT_STATUS_4	0x0F0003	/* Interrupt Status 4 */
+#define CS43130_INT_STATUS_5	0x0F0004	/* Interrupt Status 5 */
+#define CS43130_INT_MASK_1	0x0F0010        /* Interrupt Mask 1 */
+#define CS43130_INT_MASK_2	0x0F0011	/* Interrupt Mask 2 */
+#define CS43130_INT_MASK_3	0x0F0012        /* Interrupt Mask 3 */
+#define CS43130_INT_MASK_4	0x0F0013        /* Interrupt Mask 4 */
+#define CS43130_INT_MASK_5	0x0F0014        /* Interrupt Mask 5 */
+
+#define CS43130_MCLK_SRC_SEL_MASK	0x03
+#define CS43130_MCLK_SRC_SEL_SHIFT	0
+#define CS43130_MCLK_INT_MASK		0x04
+#define CS43130_MCLK_INT_SHIFT		2
+#define CS43130_CH_BITSIZE_MASK		0x03
+#define CS43130_CH_EN_MASK		0x04
+#define CS43130_CH_EN_SHIFT		2
+#define CS43130_ASP_BITSIZE_MASK	0x03
+#define CS43130_XSP_BITSIZE_MASK	0x0C
+#define CS43130_XSP_BITSIZE_SHIFT	2
+#define CS43130_SP_BITSIZE_ASP_SHIFT	0
+#define CS43130_HP_DETECT_CTRL_SHIFT	6
+#define CS43130_HP_DETECT_CTRL_MASK     (0x03 << CS43130_HP_DETECT_CTRL_SHIFT)
+#define CS43130_HP_DETECT_INV_SHIFT	5
+#define CS43130_HP_DETECT_INV_MASK      (1 << CS43130_HP_DETECT_INV_SHIFT)
+
+/* CS43130_INT_MASK_1 */
+#define CS43130_HP_PLUG_INT_SHIFT       6
+#define CS43130_HP_PLUG_INT             (1 << CS43130_HP_PLUG_INT_SHIFT)
+#define CS43130_HP_UNPLUG_INT_SHIFT     5
+#define CS43130_HP_UNPLUG_INT           (1 << CS43130_HP_UNPLUG_INT_SHIFT)
+#define CS43130_XTAL_RDY_INT_SHIFT      4
+#define CS43130_XTAL_RDY_INT_MASK	0x10
+#define CS43130_XTAL_RDY_INT            (1 << CS43130_XTAL_RDY_INT_SHIFT)
+#define CS43130_XTAL_ERR_INT_SHIFT      3
+#define CS43130_XTAL_ERR_INT            (1 << CS43130_XTAL_ERR_INT_SHIFT)
+#define CS43130_PLL_RDY_INT_MASK	0x04
+#define CS43130_PLL_RDY_INT_SHIFT	2
+#define CS43130_PLL_RDY_INT		(1 << CS43130_PLL_RDY_INT_SHIFT)
+
+/* CS43130_INT_MASK_4 */
+#define CS43130_INT_MASK_ALL		0xFF
+#define CS43130_HPLOAD_NO_DC_INT_SHIFT	7
+#define CS43130_HPLOAD_NO_DC_INT	(1 << CS43130_HPLOAD_NO_DC_INT_SHIFT)
+#define CS43130_HPLOAD_UNPLUG_INT_SHIFT	6
+#define CS43130_HPLOAD_UNPLUG_INT	(1 << CS43130_HPLOAD_UNPLUG_INT_SHIFT)
+#define CS43130_HPLOAD_OOR_INT_SHIFT	4
+#define CS43130_HPLOAD_OOR_INT		(1 << CS43130_HPLOAD_OOR_INT_SHIFT)
+#define CS43130_HPLOAD_AC_INT_SHIFT	3
+#define CS43130_HPLOAD_AC_INT		(1 << CS43130_HPLOAD_AC_INT_SHIFT)
+#define CS43130_HPLOAD_DC_INT_SHIFT	2
+#define CS43130_HPLOAD_DC_INT		(1 << CS43130_HPLOAD_DC_INT_SHIFT)
+#define CS43130_HPLOAD_OFF_INT_SHIFT	1
+#define CS43130_HPLOAD_OFF_INT		(1 << CS43130_HPLOAD_OFF_INT_SHIFT)
+#define CS43130_HPLOAD_ON_INT		1
+
+/* CS43130_HP_LOAD_1 */
+#define CS43130_HPLOAD_EN_SHIFT		7
+#define CS43130_HPLOAD_EN		(1 << CS43130_HPLOAD_EN_SHIFT)
+#define CS43130_HPLOAD_CHN_SEL_SHIFT	4
+#define CS43130_HPLOAD_CHN_SEL		(1 << CS43130_HPLOAD_CHN_SEL_SHIFT)
+#define CS43130_HPLOAD_AC_START_SHIFT	1
+#define CS43130_HPLOAD_AC_START		(1 << CS43130_HPLOAD_AC_START_SHIFT)
+#define CS43130_HPLOAD_DC_START		1
+
+/* Reg CS43130_SP_BITSIZE */
+#define CS43130_SP_BIT_SIZE_8	0x03
+#define CS43130_SP_BIT_SIZE_16	0x02
+#define CS43130_SP_BIT_SIZE_24	0x01
+#define CS43130_SP_BIT_SIZE_32	0x00
+
+/* Reg CS43130_SP_CH_SZ_EN */
+#define CS43130_CH_BIT_SIZE_8	0x00
+#define CS43130_CH_BIT_SIZE_16	0x01
+#define CS43130_CH_BIT_SIZE_24	0x02
+#define CS43130_CH_BIT_SIZE_32	0x03
+
+/* PLL */
+#define CS43130_PLL_START_MASK	0x01
+#define CS43130_PLL_MODE_MASK	0x02
+#define CS43130_PLL_MODE_SHIFT	1
+
+#define CS43130_PLL_REF_PREDIV_MASK	0x3
+
+#define CS43130_SP_STP_MASK	0x10
+#define CS43130_SP_STP_SHIFT	4
+#define CS43130_SP_5050_MASK	0x08
+#define CS43130_SP_5050_SHIFT	3
+#define CS43130_SP_FSD_MASK	0x07
+
+#define CS43130_SP_MODE_MASK	0x10
+#define CS43130_SP_MODE_SHIFT	4
+#define CS43130_SP_SCPOL_OUT_MASK	0x08
+#define CS43130_SP_SCPOL_OUT_SHIFT	3
+#define CS43130_SP_SCPOL_IN_MASK	0x04
+#define CS43130_SP_SCPOL_IN_SHIFT	2
+#define CS43130_SP_LCPOL_OUT_MASK	0x02
+#define CS43130_SP_LCPOL_OUT_SHIFT	1
+#define CS43130_SP_LCPOL_IN_MASK	0x01
+#define CS43130_SP_LCPOL_IN_SHIFT	0
+
+/* Reg CS43130_PWDN_CTL */
+#define CS43130_PDN_XSP_MASK	0x80
+#define CS43130_PDN_XSP_SHIFT	7
+#define CS43130_PDN_ASP_MASK	0x40
+#define CS43130_PDN_ASP_SHIFT	6
+#define CS43130_PDN_DSPIF_MASK	0x20
+#define CS43130_PDN_DSDIF_SHIFT	5
+#define CS43130_PDN_HP_MASK	0x10
+#define CS43130_PDN_HP_SHIFT	4
+#define CS43130_PDN_XTAL_MASK	0x08
+#define CS43130_PDN_XTAL_SHIFT	3
+#define CS43130_PDN_PLL_MASK	0x04
+#define CS43130_PDN_PLL_SHIFT	2
+#define CS43130_PDN_CLKOUT_MASK	0x02
+#define CS43130_PDN_CLKOUT_SHIFT	1
+
+/* Reg CS43130_HP_OUT_CTL_1 */
+#define CS43130_HP_IN_EN_SHIFT		3
+#define CS43130_HP_IN_EN_MASK		0x08
+
+/* Reg CS43130_PAD_INT_CFG */
+#define CS43130_ASP_3ST_MASK		0x01
+#define CS43130_XSP_3ST_MASK		0x02
+
+/* Reg CS43130_PLL_SET_2 */
+#define CS43130_PLL_DIV_DATA_MASK	0x000000FF
+#define CS43130_PLL_DIV_FRAC_0_DATA_SHIFT	0
+
+/* Reg CS43130_PLL_SET_3 */
+#define CS43130_PLL_DIV_FRAC_1_DATA_SHIFT	8
+
+/* Reg CS43130_PLL_SET_4 */
+#define CS43130_PLL_DIV_FRAC_2_DATA_SHIFT	16
+
+/* Reg CS43130_SP_DEN_1 */
+#define CS43130_SP_M_LSB_DATA_MASK	0x00FF
+#define CS43130_SP_M_LSB_DATA_SHIFT	0
+
+/* Reg CS43130_SP_DEN_2 */
+#define CS43130_SP_M_MSB_DATA_MASK	0xFF00
+#define CS43130_SP_M_MSB_DATA_SHIFT	8
+
+/* Reg CS43130_SP_NUM_1 */
+#define CS43130_SP_N_LSB_DATA_MASK	0x00FF
+#define CS43130_SP_N_LSB_DATA_SHIFT	0
+
+/* Reg CS43130_SP_NUM_2 */
+#define CS43130_SP_N_MSB_DATA_MASK	0xFF00
+#define CS43130_SP_N_MSB_DATA_SHIFT	8
+
+/* Reg CS43130_SP_LRCK_HI_TIME_1 */
+#define	CS43130_SP_LCHI_DATA_MASK	0x00FF
+#define CS43130_SP_LCHI_LSB_DATA_SHIFT	0
+
+/* Reg CS43130_SP_LRCK_HI_TIME_2 */
+#define CS43130_SP_LCHI_MSB_DATA_SHIFT	8
+
+/* Reg CS43130_SP_LRCK_PERIOD_1 */
+#define CS43130_SP_LCPR_DATA_MASK	0x00FF
+#define CS43130_SP_LCPR_LSB_DATA_SHIFT	0
+
+/* Reg CS43130_SP_LRCK_PERIOD_2 */
+#define CS43130_SP_LCPR_MSB_DATA_SHIFT	8
+
+#define CS43130_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8  | \
+			SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE)
+
+#define CS43130_DOP_FORMATS (SNDRV_PCM_FMTBIT_DSD_U16_LE | \
+			     SNDRV_PCM_FMTBIT_DSD_U16_BE | \
+			     SNDRV_PCM_FMTBIT_S24_LE)
+
+/* Reg CS43130_CRYSTAL_SET */
+#define CS43130_XTAL_IBIAS_MASK		0x07
+
+/* Reg CS43130_PATH_CTL_1 */
+#define CS43130_MUTE_MASK		0x03
+#define CS43130_MUTE_EN			0x03
+
+/* Reg CS43130_DSD_INT_CFG */
+#define CS43130_DSD_MASTER		0x04
+
+/* Reg CS43130_DSD_PATH_CTL_2 */
+#define CS43130_DSD_SRC_MASK		0x60
+#define CS43130_DSD_SRC_SHIFT		5
+#define CS43130_DSD_EN_SHIFT		4
+#define CS43130_DSD_SPEED_MASK		0x04
+#define CS43130_DSD_SPEED_SHIFT		2
+
+/* Reg CS43130_DSD_PCM_MIX_CTL	*/
+#define CS43130_MIX_PCM_PREP_SHIFT	1
+#define CS43130_MIX_PCM_PREP_MASK	0x02
+
+#define CS43130_MIX_PCM_DSD_SHIFT	0
+#define CS43130_MIX_PCM_DSD_MASK	0x01
+
+/* Reg CS43130_HP_MEAS_LOAD */
+#define CS43130_HP_MEAS_LOAD_MASK	0x000000FF
+#define CS43130_HP_MEAS_LOAD_1_SHIFT	0
+#define CS43130_HP_MEAS_LOAD_2_SHIFT	8
+
+#define CS43130_MCLK_22M		22579200
+#define CS43130_MCLK_24M		24576000
+
+#define CS43130_LINEOUT_LOAD		5000
+#define CS43130_JACK_LINEOUT		(SND_JACK_MECHANICAL | SND_JACK_LINEOUT)
+#define CS43130_JACK_HEADPHONE		(SND_JACK_MECHANICAL | \
+					 SND_JACK_HEADPHONE)
+#define CS43130_JACK_MASK		(SND_JACK_MECHANICAL | \
+					 SND_JACK_LINEOUT | \
+					 SND_JACK_HEADPHONE)
+
+enum cs43130_dsd_src {
+	CS43130_DSD_SRC_DSD = 0,
+	CS43130_DSD_SRC_ASP = 2,
+	CS43130_DSD_SRC_XSP = 3,
+};
+
+enum cs43130_asp_rate {
+	CS43130_ASP_SPRATE_32K = 0,
+	CS43130_ASP_SPRATE_44_1K,
+	CS43130_ASP_SPRATE_48K,
+	CS43130_ASP_SPRATE_88_2K,
+	CS43130_ASP_SPRATE_96K,
+	CS43130_ASP_SPRATE_176_4K,
+	CS43130_ASP_SPRATE_192K,
+	CS43130_ASP_SPRATE_352_8K,
+	CS43130_ASP_SPRATE_384K,
+};
+
+enum cs43130_mclk_src_sel {
+	CS43130_MCLK_SRC_EXT = 0,
+	CS43130_MCLK_SRC_PLL,
+	CS43130_MCLK_SRC_RCO
+};
+
+enum cs43130_mclk_int_freq {
+	CS43130_MCLK_24P5 = 0,
+	CS43130_MCLK_22P5,
+};
+
+enum cs43130_xtal_ibias {
+	CS43130_XTAL_UNUSED = -1,
+	CS43130_XTAL_IBIAS_15UA = 2,
+	CS43130_XTAL_IBIAS_12_5UA = 4,
+	CS43130_XTAL_IBIAS_7_5UA = 6,
+};
+
+enum cs43130_dai_id {
+	CS43130_ASP_PCM_DAI = 0,
+	CS43130_ASP_DOP_DAI,
+	CS43130_XSP_DOP_DAI,
+	CS43130_XSP_DSD_DAI,
+	CS43130_DAI_ID_MAX,
+};
+
+struct cs43130_clk_gen {
+	unsigned int	mclk_int;
+	int		fs;
+	u16		den;
+	u16		num;
+};
+
+/* frm_size = 16 */
+static const struct cs43130_clk_gen cs43130_16_clk_gen[] = {
+	{22579200,	32000,		441,		10,},
+	{22579200,	44100,		32,		1,},
+	{22579200,	48000,		147,		5,},
+	{22579200,	88200,		16,		1,},
+	{22579200,	96000,		147,		10,},
+	{22579200,	176400,		8,		1,},
+	{22579200,	192000,		147,		20,},
+	{22579200,	352800,		4,		1,},
+	{22579200,	384000,		147,		40,},
+	{24576000,	32000,		48,		1,},
+	{24576000,	44100,		5120,		147,},
+	{24576000,	48000,		32,		1,},
+	{24576000,	88200,		2560,		147,},
+	{24576000,	96000,		16,		1,},
+	{24576000,	176400,		1280,		147,},
+	{24576000,	192000,		8,		1,},
+	{24576000,	352800,		640,		147,},
+	{24576000,	384000,		4,		1,},
+};
+
+/* frm_size = 32 */
+static const struct cs43130_clk_gen cs43130_32_clk_gen[] = {
+	{22579200,	32000,		441,		20,},
+	{22579200,	44100,		16,		1,},
+	{22579200,	48000,		147,		10,},
+	{22579200,	88200,		8,		1,},
+	{22579200,	96000,		147,		20,},
+	{22579200,	176400,		4,		1,},
+	{22579200,	192000,		147,		40,},
+	{22579200,	352800,		2,		1,},
+	{22579200,	384000,		147,		80,},
+	{24576000,	32000,		24,		1,},
+	{24576000,	44100,		2560,		147,},
+	{24576000,	48000,		16,		1,},
+	{24576000,	88200,		1280,		147,},
+	{24576000,	96000,		8,		1,},
+	{24576000,	176400,		640,		147,},
+	{24576000,	192000,		4,		1,},
+	{24576000,	352800,		320,		147,},
+	{24576000,	384000,		2,		1,},
+};
+
+/* frm_size = 48 */
+static const struct cs43130_clk_gen cs43130_48_clk_gen[] = {
+	{22579200,	32000,		147,		100,},
+	{22579200,	44100,		32,		3,},
+	{22579200,	48000,		49,		5,},
+	{22579200,	88200,		16,		3,},
+	{22579200,	96000,		49,		10,},
+	{22579200,	176400,		8,		3,},
+	{22579200,	192000,		49,		20,},
+	{22579200,	352800,		4,		3,},
+	{22579200,	384000,		49,		40,},
+	{24576000,	32000,		16,		1,},
+	{24576000,	44100,		5120,		441,},
+	{24576000,	48000,		32,		3,},
+	{24576000,	88200,		2560,		441,},
+	{24576000,	96000,		16,		3,},
+	{24576000,	176400,		1280,		441,},
+	{24576000,	192000,		8,		3,},
+	{24576000,	352800,		640,		441,},
+	{24576000,	384000,		4,		3,},
+};
+
+/* frm_size = 64 */
+static const struct cs43130_clk_gen cs43130_64_clk_gen[] = {
+	{22579200,	32000,		441,		40,},
+	{22579200,	44100,		8,		1,},
+	{22579200,	48000,		147,		20,},
+	{22579200,	88200,		4,		1,},
+	{22579200,	96000,		147,		40,},
+	{22579200,	176400,		2,		1,},
+	{22579200,	192000,		147,		80,},
+	{22579200,	352800,		1,		1,},
+	{24576000,	32000,		12,		1,},
+	{24576000,	44100,		1280,		147,},
+	{24576000,	48000,		8,		1,},
+	{24576000,	88200,		640,		147,},
+	{24576000,	96000,		4,		1,},
+	{24576000,	176400,		320,		147,},
+	{24576000,	192000,		2,		1,},
+	{24576000,	352800,		160,		147,},
+	{24576000,	384000,		1,		1,},
+};
+
+struct cs43130_bitwidth_map {
+	unsigned int bitwidth;
+	u8 sp_bit;
+	u8 ch_bit;
+};
+
+struct cs43130_rate_map {
+	int fs;
+	int val;
+};
+
+#define HP_LEFT			0
+#define HP_RIGHT		1
+#define CS43130_AC_FREQ		10
+#define CS43130_DC_THRESHOLD	2
+
+#define CS43130_NUM_SUPPLIES	5
+static const char *const cs43130_supply_names[CS43130_NUM_SUPPLIES] = {
+	"VA",
+	"VP",
+	"VCP",
+	"VD",
+	"VL",
+};
+
+#define CS43130_NUM_INT		5       /* number of interrupt status reg */
+
+struct cs43130_dai {
+	unsigned int			sclk;
+	unsigned int			dai_format;
+	unsigned int			dai_mode;
+};
+
+struct	cs43130_private {
+	struct snd_soc_codec		*codec;
+	struct regmap			*regmap;
+	struct regulator_bulk_data	supplies[CS43130_NUM_SUPPLIES];
+	struct gpio_desc		*reset_gpio;
+	unsigned int			dev_id; /* codec device ID */
+	int				xtal_ibias;
+
+	/* shared by both DAIs */
+	struct mutex			clk_mutex;
+	int				clk_req;
+	bool				pll_bypass;
+	struct completion		xtal_rdy;
+	struct completion		pll_rdy;
+	unsigned int			mclk;
+	unsigned int			mclk_int;
+	int				mclk_int_src;
+
+	/* DAI specific */
+	struct cs43130_dai		dais[CS43130_DAI_ID_MAX];
+
+	/* HP load specific */
+	bool				dc_meas;
+	bool				ac_meas;
+	bool				hpload_done;
+	struct completion		hpload_evt;
+	unsigned int			hpload_stat;
+	u16				hpload_dc[2];
+	u16				dc_threshold[CS43130_DC_THRESHOLD];
+	u16				ac_freq[CS43130_AC_FREQ];
+	u16				hpload_ac[CS43130_AC_FREQ][2];
+	struct workqueue_struct		*wq;
+	struct work_struct		work;
+	struct snd_soc_jack		jack;
+};
+
+#endif	/* __CS43130_H__ */
-- 
cgit v1.2.3


From a3dcadd220b4572f9baaad5f3c0c00044273c961 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Fri, 1 Sep 2017 04:33:01 +0000
Subject: ASoC: add missing compile rule for max98371

It is still using old driver style, this patch also
fixup it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Makefile   |  1 +
 sound/soc/codecs/max98371.c | 14 ++++++++------
 2 files changed, 9 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1755a54e3dc9..b4d576dab48a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -320,6 +320,7 @@ obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98090)	+= snd-soc-max98090.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
 obj-$(CONFIG_SND_SOC_MAX98357A)	+= snd-soc-max98357a.o
+obj-$(CONFIG_SND_SOC_MAX98371)	+= snd-soc-max98371.o
 obj-$(CONFIG_SND_SOC_MAX9867)	+= snd-soc-max9867.o
 obj-$(CONFIG_SND_SOC_MAX98925)	+= snd-soc-max98925.o
 obj-$(CONFIG_SND_SOC_MAX98926)	+= snd-soc-max98926.o
diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
index 781be9ba8dba..7bc2a17c1e94 100644
--- a/sound/soc/codecs/max98371.c
+++ b/sound/soc/codecs/max98371.c
@@ -349,12 +349,14 @@ static struct snd_soc_dai_driver max98371_dai[] = {
 };
 
 static const struct snd_soc_codec_driver max98371_codec = {
-	.controls = max98371_snd_controls,
-	.num_controls = ARRAY_SIZE(max98371_snd_controls),
-	.dapm_routes = max98371_audio_map,
-	.num_dapm_routes = ARRAY_SIZE(max98371_audio_map),
-	.dapm_widgets = max98371_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets),
+	.component_driver = {
+		.controls = max98371_snd_controls,
+		.num_controls = ARRAY_SIZE(max98371_snd_controls),
+		.dapm_routes = max98371_audio_map,
+		.num_dapm_routes = ARRAY_SIZE(max98371_audio_map),
+		.dapm_widgets = max98371_dapm_widgets,
+		.num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets),
+	},
 };
 
 static const struct regmap_config max98371_regmap = {
-- 
cgit v1.2.3


From 2dc3b7b06bace6d8986a5fae6569ef30276cdd73 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Fri, 1 Sep 2017 04:33:27 +0000
Subject: ASoC: add missing compile rule for sirf-audio-codec

It is still using old driver style, this patch also
fixup it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Makefile           |  1 +
 sound/soc/codecs/sirf-audio-codec.c | 10 ++++++----
 2 files changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index b4d576dab48a..8ed3c9256a74 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -373,6 +373,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP)	+= snd-soc-sigmadsp-regmap.o
 obj-$(CONFIG_SND_SOC_SI476X)	+= snd-soc-si476x.o
 obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif-rx.o snd-soc-spdif-tx.o
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o
 obj-$(CONFIG_SND_SOC_SSM2518)	+= snd-soc-ssm2518.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_SSM2602_SPI)	+= snd-soc-ssm2602-spi.o
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
index 6bfd25c289d1..01d7690a1520 100644
--- a/sound/soc/codecs/sirf-audio-codec.c
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -432,10 +432,12 @@ static int sirf_audio_codec_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
 	.probe = sirf_audio_codec_probe,
 	.remove = sirf_audio_codec_remove,
-	.dapm_widgets = sirf_audio_codec_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets),
-	.dapm_routes = sirf_audio_codec_map,
-	.num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map),
+	.component_driver = {
+		.dapm_widgets = sirf_audio_codec_dapm_widgets,
+		.num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets),
+		.dapm_routes = sirf_audio_codec_map,
+		.num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map),
+	},
 	.idle_bias_off = true,
 };
 
-- 
cgit v1.2.3


From b21f0522ee376ddbea54fab47ff36f687f2a438b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto
Date: Fri, 1 Sep 2017 04:33:01 +0000
Subject: ASoC: add missing compile rule for max98371

It is still using old driver style, this patch also
fixup it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Makefile   |  1 +
 sound/soc/codecs/max98371.c | 14 ++++++++------
 2 files changed, 9 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1755a54e3dc9..b4d576dab48a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -320,6 +320,7 @@ obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98090)	+= snd-soc-max98090.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
 obj-$(CONFIG_SND_SOC_MAX98357A)	+= snd-soc-max98357a.o
+obj-$(CONFIG_SND_SOC_MAX98371)	+= snd-soc-max98371.o
 obj-$(CONFIG_SND_SOC_MAX9867)	+= snd-soc-max9867.o
 obj-$(CONFIG_SND_SOC_MAX98925)	+= snd-soc-max98925.o
 obj-$(CONFIG_SND_SOC_MAX98926)	+= snd-soc-max98926.o
diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
index 781be9ba8dba..7bc2a17c1e94 100644
--- a/sound/soc/codecs/max98371.c
+++ b/sound/soc/codecs/max98371.c
@@ -349,12 +349,14 @@ static struct snd_soc_dai_driver max98371_dai[] = {
 };
 
 static const struct snd_soc_codec_driver max98371_codec = {
-	.controls = max98371_snd_controls,
-	.num_controls = ARRAY_SIZE(max98371_snd_controls),
-	.dapm_routes = max98371_audio_map,
-	.num_dapm_routes = ARRAY_SIZE(max98371_audio_map),
-	.dapm_widgets = max98371_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets),
+	.component_driver = {
+		.controls = max98371_snd_controls,
+		.num_controls = ARRAY_SIZE(max98371_snd_controls),
+		.dapm_routes = max98371_audio_map,
+		.num_dapm_routes = ARRAY_SIZE(max98371_audio_map),
+		.dapm_widgets = max98371_dapm_widgets,
+		.num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets),
+	},
 };
 
 static const struct regmap_config max98371_regmap = {
-- 
cgit v1.2.3


From 38a770859e10f566c434ae0d3f380954280f262b Mon Sep 17 00:00:00 2001
From: Pradeep Tewani
Date: Fri, 1 Sep 2017 13:36:13 +0530
Subject: ASoC: Intel: Skylake: Add IPC to configure the copier secondary pins

Copier can support upto 4 output pins. However, only pin 0 is configured
as a part of copier initialization. Configuring rest of pins require the
separate IPC to be sent to fw.

Signed-off-by: Pradeep Tewani <pradeep.d.tewani@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 53 ++++++++++++++++++++++++++++++----
 sound/soc/intel/skylake/skl-topology.h |  5 ++++
 2 files changed, 53 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index ea83e8bfb136..89f70133c8e4 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -1083,6 +1083,21 @@ int skl_unbind_modules(struct skl_sst *ctx,
 	return ret;
 }
 
+static void fill_pin_params(struct skl_audio_data_format *pin_fmt,
+				struct skl_module_fmt *format)
+{
+	pin_fmt->number_of_channels = format->channels;
+	pin_fmt->s_freq = format->s_freq;
+	pin_fmt->bit_depth = format->bit_depth;
+	pin_fmt->valid_bit_depth = format->valid_bit_depth;
+	pin_fmt->ch_cfg = format->ch_cfg;
+	pin_fmt->sample_type = format->sample_type;
+	pin_fmt->channel_map = format->ch_map;
+	pin_fmt->interleaving = format->interleaving_style;
+}
+
+#define CPR_SINK_FMT_PARAM_ID 2
+
 /*
  * Once a module is instantiated it need to be 'bind' with other modules in
  * the pipeline. For binding we need to find the module pins which are bind
@@ -1094,11 +1109,15 @@ int skl_bind_modules(struct skl_sst *ctx,
 			struct skl_module_cfg *src_mcfg,
 			struct skl_module_cfg *dst_mcfg)
 {
-	int ret;
+	int ret = 0;
 	struct skl_ipc_bind_unbind_msg msg;
 	int in_max = dst_mcfg->module->max_input_pins;
 	int out_max = src_mcfg->module->max_output_pins;
 	int src_index, dst_index;
+	struct skl_module_fmt *format;
+	struct skl_cpr_pin_fmt pin_fmt;
+	struct skl_module *module;
+	struct skl_module_iface *fmt;
 
 	skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
 
@@ -1117,6 +1136,29 @@ int skl_bind_modules(struct skl_sst *ctx,
 		return -EINVAL;
 	}
 
+	/*
+	 * Copier module requires the separate large_config_set_ipc to
+	 * configure the pins other than 0
+	 */
+	if (src_mcfg->m_type == SKL_MODULE_TYPE_COPIER && src_index > 0) {
+		pin_fmt.sink_id = src_index;
+		module = src_mcfg->module;
+		fmt = &module->formats[src_mcfg->fmt_idx];
+
+		/* Input fmt is same as that of src module input cfg */
+		format = &fmt->inputs[0].fmt;
+		fill_pin_params(&(pin_fmt.src_fmt), format);
+
+		format = &fmt->outputs[src_index].fmt;
+		fill_pin_params(&(pin_fmt.dst_fmt), format);
+		ret = skl_set_module_params(ctx, (void *)&pin_fmt,
+					sizeof(struct skl_cpr_pin_fmt),
+					CPR_SINK_FMT_PARAM_ID, src_mcfg);
+
+		if (ret < 0)
+			goto out;
+	}
+
 	msg.dst_queue = dst_index;
 
 	dev_dbg(ctx->dev, "src queue = %d dst queue =%d\n",
@@ -1134,11 +1176,12 @@ int skl_bind_modules(struct skl_sst *ctx,
 		src_mcfg->m_state = SKL_MODULE_BIND_DONE;
 		src_mcfg->m_out_pin[src_index].pin_state = SKL_PIN_BIND_DONE;
 		dst_mcfg->m_in_pin[dst_index].pin_state = SKL_PIN_BIND_DONE;
-	} else {
-		/* error case , if IPC fails, clear the queue index */
-		skl_free_queue(src_mcfg->m_out_pin, src_index);
-		skl_free_queue(dst_mcfg->m_in_pin, dst_index);
+		return ret;
 	}
+out:
+	/* error case , if IPC fails, clear the queue index */
+	skl_free_queue(src_mcfg->m_out_pin, src_index);
+	skl_free_queue(dst_mcfg->m_in_pin, dst_index);
 
 	return ret;
 }
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 4975904480e9..2717db92036b 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -138,6 +138,11 @@ struct skl_cpr_cfg {
 	struct skl_cpr_gtw_cfg gtw_cfg;
 } __packed;
 
+struct skl_cpr_pin_fmt {
+	u32 sink_id;
+	struct skl_audio_data_format src_fmt;
+	struct skl_audio_data_format dst_fmt;
+} __packed;
 
 struct skl_src_module_cfg {
 	struct skl_base_cfg base_cfg;
-- 
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