From 368494093354ac613a80c2e1d77602aa12473cf0 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Mon, 10 Nov 2014 12:27:33 +0200
Subject: ASoC: tlv320aic3x: Add TDM support

TDM support is achieved using DSP transfer mode and setting a
programmable offset which specifies where data begins with
respect to the frame sync.

It requires 256-clock mode if CODEC is master (not currently
supported in the driver). No additional dependency if CODEC
is slave.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic3x.c | 62 ++++++++++++++++++++++++++++++++++++++++--
 sound/soc/codecs/tlv320aic3x.h |  1 +
 2 files changed, 60 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index f7c2a575a892..8770e28e53a4 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -78,6 +78,8 @@ struct aic3x_priv {
 	struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES];
 	struct aic3x_setup_data *setup;
 	unsigned int sysclk;
+	unsigned int dai_fmt;
+	unsigned int tdm_delay;
 	struct list_head list;
 	int master;
 	int gpio_reset;
@@ -1009,6 +1011,25 @@ found:
 	return 0;
 }
 
+static int aic3x_prepare(struct snd_pcm_substream *substream,
+			 struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+	int delay = 0;
+
+	/* TDM slot selection only valid in DSP_A/_B mode */
+	if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
+		delay += (aic3x->tdm_delay + 1);
+	else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
+		delay += aic3x->tdm_delay;
+
+	/* Configure data delay */
+	snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay);
+
+	return 0;
+}
+
 static int aic3x_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
@@ -1048,7 +1069,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	u8 iface_areg, iface_breg;
-	int delay = 0;
 
 	iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
 	iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
@@ -1076,7 +1096,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
 		break;
 	case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF):
-		delay = 1;
 	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
 		iface_breg |= (0x01 << 6);
 		break;
@@ -1090,10 +1109,45 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
+	aic3x->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
 	/* set iface */
 	snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
 	snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
-	snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
+
+	return 0;
+}
+
+static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
+				  unsigned int tx_mask, unsigned int rx_mask,
+				  int slots, int slot_width)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+	unsigned int lsb;
+
+	if (tx_mask != rx_mask) {
+		dev_err(codec->dev, "tx and rx masks must be symmetric\n");
+		return -EINVAL;
+	}
+
+	if (unlikely(!tx_mask)) {
+		dev_err(codec->dev, "tx and rx masks need to be non 0\n");
+		return -EINVAL;
+	}
+
+	/* TDM based on DSP mode requires slots to be adjacent */
+	lsb = __ffs(tx_mask);
+	if ((lsb + 1) != __fls(tx_mask)) {
+		dev_err(codec->dev, "Invalid mask, slots must be adjacent\n");
+		return -EINVAL;
+	}
+
+	aic3x->tdm_delay = lsb * slot_width;
+
+	/* DOUT in high-impedance on inactive bit clocks */
+	snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA,
+			    DOUT_TRISTATE, DOUT_TRISTATE);
 
 	return 0;
 }
@@ -1212,9 +1266,11 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 
 static const struct snd_soc_dai_ops aic3x_dai_ops = {
 	.hw_params	= aic3x_hw_params,
+	.prepare	= aic3x_prepare,
 	.digital_mute	= aic3x_mute,
 	.set_sysclk	= aic3x_set_dai_sysclk,
 	.set_fmt	= aic3x_set_dai_fmt,
+	.set_tdm_slot	= aic3x_set_dai_tdm_slot,
 };
 
 static struct snd_soc_dai_driver aic3x_dai = {
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index e521ac3ddde8..89fa692df206 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -169,6 +169,7 @@
 /* Audio serial data interface control register A bits */
 #define BIT_CLK_MASTER          0x80
 #define WORD_CLK_MASTER         0x40
+#define DOUT_TRISTATE		0x20
 
 /* Codec Datapath setup register 7 */
 #define FSREF_44100		(1 << 7)
-- 
cgit v1.2.3-58-ga151


From a60e654be733a69879148cb4c56d0f58b749e3c4 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Tue, 11 Nov 2014 10:59:00 +0200
Subject: ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to
 individual

It is easier to find the relevant enums in the code. Use the
SOC_ENUM_*_DECL macro for the individual items.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic3x.c | 151 +++++++++++++++++++++--------------------
 1 file changed, 79 insertions(+), 72 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8770e28e53a4..990140058aa6 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -216,61 +216,68 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" };
-static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" };
-static const char *aic3x_left_hpcom_mux[] =
-    { "differential of HPLOUT", "constant VCM", "single-ended" };
-static const char *aic3x_right_hpcom_mux[] =
-    { "differential of HPROUT", "constant VCM", "single-ended",
-      "differential of HPLCOM", "external feedback" };
-static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
-static const char *aic3x_adc_hpf[] =
-    { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
-
-#define LDAC_ENUM	0
-#define RDAC_ENUM	1
-#define LHPCOM_ENUM	2
-#define RHPCOM_ENUM	3
-#define LINE1L_2_L_ENUM	4
-#define LINE1L_2_R_ENUM	5
-#define LINE1R_2_L_ENUM	6
-#define LINE1R_2_R_ENUM	7
-#define LINE2L_ENUM	8
-#define LINE2R_ENUM	9
-#define ADC_HPF_ENUM	10
-
-static const struct soc_enum aic3x_enum[] = {
-	SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
-	SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux),
-	SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux),
-	SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux),
-	SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
-	SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
-	SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
-	SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
-	SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
-	SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
-	SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
-};
-
-static const char *aic3x_agc_level[] =
-	{ "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" };
-static const struct soc_enum aic3x_agc_level_enum[] = {
-	SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level),
-	SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level),
-};
-
-static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" };
-static const struct soc_enum aic3x_agc_attack_enum[] = {
-	SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack),
-	SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack),
-};
-
-static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" };
-static const struct soc_enum aic3x_agc_decay_enum[] = {
-	SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay),
-	SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
-};
+static const char * const aic3x_left_dac_mux[] = {
+	"DAC_L1", "DAC_L3", "DAC_L2" };
+static SOC_ENUM_SINGLE_DECL(aic3x_left_dac_enum, DAC_LINE_MUX, 6,
+			    aic3x_left_dac_mux);
+
+static const char * const aic3x_right_dac_mux[] = {
+	"DAC_R1", "DAC_R3", "DAC_R2" };
+static SOC_ENUM_SINGLE_DECL(aic3x_right_dac_enum, DAC_LINE_MUX, 4,
+			    aic3x_right_dac_mux);
+
+static const char * const aic3x_left_hpcom_mux[] = {
+	"differential of HPLOUT", "constant VCM", "single-ended" };
+static SOC_ENUM_SINGLE_DECL(aic3x_left_hpcom_enum, HPLCOM_CFG, 4,
+			    aic3x_left_hpcom_mux);
+
+static const char * const aic3x_right_hpcom_mux[] = {
+	"differential of HPROUT", "constant VCM", "single-ended",
+	"differential of HPLCOM", "external feedback" };
+static SOC_ENUM_SINGLE_DECL(aic3x_right_hpcom_enum, HPRCOM_CFG, 3,
+			    aic3x_right_hpcom_mux);
+
+static const char * const aic3x_linein_mode_mux[] = {
+	"single-ended", "differential" };
+static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_l_enum, LINE1L_2_LADC_CTRL, 7,
+			    aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_r_enum, LINE1L_2_RADC_CTRL, 7,
+			    aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_l_enum, LINE1R_2_LADC_CTRL, 7,
+			    aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_r_enum, LINE1R_2_RADC_CTRL, 7,
+			    aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line2l_2_ldac_enum, LINE2L_2_LADC_CTRL, 7,
+			    aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line2r_2_rdac_enum, LINE2R_2_RADC_CTRL, 7,
+			    aic3x_linein_mode_mux);
+
+static const char * const aic3x_adc_hpf[] = {
+	"Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
+static SOC_ENUM_DOUBLE_DECL(aic3x_adc_hpf_enum, AIC3X_CODEC_DFILT_CTRL, 6, 4,
+			    aic3x_adc_hpf);
+
+static const char * const aic3x_agc_level[] = {
+	"-5.5dB", "-8dB", "-10dB", "-12dB",
+	"-14dB", "-17dB", "-20dB", "-24dB" };
+static SOC_ENUM_SINGLE_DECL(aic3x_lagc_level_enum, LAGC_CTRL_A, 4,
+			    aic3x_agc_level);
+static SOC_ENUM_SINGLE_DECL(aic3x_ragc_level_enum, RAGC_CTRL_A, 4,
+			    aic3x_agc_level);
+
+static const char * const aic3x_agc_attack[] = {
+	"8ms", "11ms", "16ms", "20ms" };
+static SOC_ENUM_SINGLE_DECL(aic3x_lagc_attack_enum, LAGC_CTRL_A, 2,
+			    aic3x_agc_attack);
+static SOC_ENUM_SINGLE_DECL(aic3x_ragc_attack_enum, RAGC_CTRL_A, 2,
+			    aic3x_agc_attack);
+
+static const char * const aic3x_agc_decay[] = {
+	"100ms", "200ms", "400ms", "500ms" };
+static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0,
+			    aic3x_agc_decay);
+static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0,
+			    aic3x_agc_decay);
 
 /*
  * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
@@ -385,12 +392,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	 * adjust PGA to max value when ADC is on and will never go back.
 	*/
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
-	SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]),
-	SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]),
-	SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]),
-	SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]),
-	SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]),
-	SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]),
+	SOC_ENUM("Left AGC Target level", aic3x_lagc_level_enum),
+	SOC_ENUM("Right AGC Target level", aic3x_ragc_level_enum),
+	SOC_ENUM("Left AGC Attack time", aic3x_lagc_attack_enum),
+	SOC_ENUM("Right AGC Attack time", aic3x_ragc_attack_enum),
+	SOC_ENUM("Left AGC Decay time", aic3x_lagc_decay_enum),
+	SOC_ENUM("Right AGC Decay time", aic3x_ragc_decay_enum),
 
 	/* De-emphasis */
 	SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0),
@@ -400,7 +407,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 			 0, 119, 0, adc_tlv),
 	SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
 
-	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
+	SOC_ENUM("ADC HPF Cut-off", aic3x_adc_hpf_enum),
 };
 
 static const struct snd_kcontrol_new aic3x_mono_controls[] = {
@@ -427,19 +434,19 @@ static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
 
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_left_dac_enum);
 
 /* Right DAC Mux */
 static const struct snd_kcontrol_new aic3x_right_dac_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_right_dac_enum);
 
 /* Left HPCOM Mux */
 static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_left_hpcom_enum);
 
 /* Right HPCOM Mux */
 static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum);
 
 /* Left Line Mixer */
 static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = {
@@ -531,23 +538,23 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
 
 /* Left Line1 Mux */
 static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum);
 static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1l_2_r_enum);
 
 /* Right Line1 Mux */
 static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1r_2_r_enum);
 static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1r_2_l_enum);
 
 /* Left Line2 Mux */
 static const struct snd_kcontrol_new aic3x_left_line2_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line2l_2_ldac_enum);
 
 /* Right Line2 Mux */
 static const struct snd_kcontrol_new aic3x_right_line2_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line2r_2_rdac_enum);
 
 static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	/* Left DAC to Left Outputs */
-- 
cgit v1.2.3-58-ga151


From 68d6626925c3529790a2055d41578415fa98495e Mon Sep 17 00:00:00 2001
From: Misael Lopez Cruz <misael.lopez@ti.com>
Date: Tue, 11 Nov 2014 10:59:01 +0200
Subject: ASoC: tlv320aic3x: Add output driver pop reduction controls

Output driver has two parameters that can be configured to reduce
pop noise: power-on delay and ramp-up step time. Two new kcontrols
have been added to set these parameters.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic3x.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 990140058aa6..f0a828119aba 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -279,6 +279,16 @@ static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0,
 static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0,
 			    aic3x_agc_decay);
 
+static const char * const aic3x_poweron_time[] = {
+	"0us", "10us", "100us", "1ms", "10ms", "50ms",
+	"100ms", "200ms", "400ms", "800ms", "2s", "4s" };
+static SOC_ENUM_SINGLE_DECL(aic3x_poweron_time_enum, HPOUT_POP_REDUCTION, 4,
+			    aic3x_poweron_time);
+
+static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" };
+static SOC_ENUM_SINGLE_DECL(aic3x_rampup_step_enum, HPOUT_POP_REDUCTION, 2,
+			    aic3x_rampup_step);
+
 /*
  * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
  */
@@ -408,6 +418,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
 
 	SOC_ENUM("ADC HPF Cut-off", aic3x_adc_hpf_enum),
+
+	/* Pop reduction */
+	SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum),
+	SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum),
 };
 
 static const struct snd_kcontrol_new aic3x_mono_controls[] = {
-- 
cgit v1.2.3-58-ga151


From 2880fc877971d6c14b0c76ac09744e3ff5b126d5 Mon Sep 17 00:00:00 2001
From: Dylan Reid <dgreid@chromium.org>
Date: Thu, 13 Nov 2014 11:18:29 -0800
Subject: ASoC: add TI ts3a227e headset chip driver

The TS3A227E is an autonomous audio accessory detection and
configuration switch that detects 3-pole or 4-pole audio accessories
and configures internal switches to route the signals accordingly.

This chip also has built-in support for the new button standard
described in the Android "Wired audio headset specification" v1.0.
These buttons will be reported on the jack as buttons 0-3 mapped to
KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND.

This will be added as an aux_dev and have the jack passed in from the
machine driver.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/ts3a227e.txt         |  26 ++
 sound/soc/codecs/Kconfig                           |   5 +
 sound/soc/codecs/Makefile                          |   2 +
 sound/soc/codecs/ts3a227e.c                        | 314 +++++++++++++++++++++
 sound/soc/codecs/ts3a227e.h                        |  17 ++
 5 files changed, 364 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt
 create mode 100644 sound/soc/codecs/ts3a227e.c
 create mode 100644 sound/soc/codecs/ts3a227e.h

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt
new file mode 100644
index 000000000000..e8bf23eb1803
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt
@@ -0,0 +1,26 @@
+Texas Instruments TS3A227E
+Autonomous Audio Accessory Detection and Configuration Switch
+
+The TS3A227E detect headsets of 3-ring and 4-ring standards and
+switches automatically to route the microphone correctly.  It also
+handles key press detection in accordance with the Android audio
+headset specification v1.0.
+
+Required properties:
+
+ - compatible:		Should contain "ti,ts3a227e".
+ - reg:			The i2c address. Should contain <0x3b>.
+ - interrupt-parent:	The parent interrupt controller
+ - interrupts:		Interrupt number for /INT pin from the 227e
+
+
+Examples:
+
+	i2c {
+		ts3a227e@3b {
+			compatible = "ti,ts3a227e";
+			reg = <0x3b>;
+			interrupt-parent = <&gpio>;
+			interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+		};
+	};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a68d1731a8fd..243ec862c426 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
+	select SND_SOC_TS3A227E if I2C
 	select SND_SOC_TWL4030 if TWL4030_CORE
 	select SND_SOC_TWL6040 if TWL6040_CORE
 	select SND_SOC_UDA134X
@@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X
 config SND_SOC_TLV320DAC33
 	tristate
 
+config SND_SOC_TS3A227E
+	tristate "TI Headset/Mic detect and keypress chip"
+	depends on I2C
+
 config SND_SOC_TWL4030
 	select MFD_TWL4030_AUDIO
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5dce451661e4..a1eb7ef3b90e 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
 snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
+snd-soc-ts3a227e-objs := ts3a227e.o
 snd-soc-twl4030-objs := twl4030.o
 snd-soc-twl6040-objs := twl6040.o
 snd-soc-uda134x-objs := uda134x.o
@@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX)     += snd-soc-tlv320aic31xx.o
 obj-$(CONFIG_SND_SOC_TLV320AIC32X4)     += snd-soc-tlv320aic32x4.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
+obj-$(CONFIG_SND_SOC_TS3A227E)	+= snd-soc-ts3a227e.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
new file mode 100644
index 000000000000..1d1205702d23
--- /dev/null
+++ b/sound/soc/codecs/ts3a227e.c
@@ -0,0 +1,314 @@
+/*
+ * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch
+ *
+ * Copyright (C) 2014 Google, Inc.
+ *
+ * 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/gpio.h>
+#include <linux/i2c.h>
+#include <linux/init.h>
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/of_gpio.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+
+struct ts3a227e {
+	struct regmap *regmap;
+	struct snd_soc_jack *jack;
+	bool plugged;
+	bool mic_present;
+	unsigned int buttons_held;
+};
+
+/* Button values to be reported on the jack */
+static const int ts3a227e_buttons[] = {
+	SND_JACK_BTN_0,
+	SND_JACK_BTN_1,
+	SND_JACK_BTN_2,
+	SND_JACK_BTN_3,
+};
+
+#define TS3A227E_NUM_BUTTONS 4
+#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \
+			    SND_JACK_MICROPHONE | \
+			    SND_JACK_BTN_0 | \
+			    SND_JACK_BTN_1 | \
+			    SND_JACK_BTN_2 | \
+			    SND_JACK_BTN_3)
+
+/* TS3A227E registers */
+#define TS3A227E_REG_DEVICE_ID		0x00
+#define TS3A227E_REG_INTERRUPT		0x01
+#define TS3A227E_REG_KP_INTERRUPT	0x02
+#define TS3A227E_REG_INTERRUPT_DISABLE	0x03
+#define TS3A227E_REG_SETTING_1		0x04
+#define TS3A227E_REG_SETTING_2		0x05
+#define TS3A227E_REG_SETTING_3		0x06
+#define TS3A227E_REG_SWITCH_CONTROL_1	0x07
+#define TS3A227E_REG_SWITCH_CONTROL_2	0x08
+#define TS3A227E_REG_SWITCH_STATUS_1	0x09
+#define TS3A227E_REG_SWITCH_STATUS_2	0x0a
+#define TS3A227E_REG_ACCESSORY_STATUS	0x0b
+#define TS3A227E_REG_ADC_OUTPUT		0x0c
+#define TS3A227E_REG_KP_THRESHOLD_1	0x0d
+#define TS3A227E_REG_KP_THRESHOLD_2	0x0e
+#define TS3A227E_REG_KP_THRESHOLD_3	0x0f
+
+/* TS3A227E_REG_INTERRUPT 0x01 */
+#define INS_REM_EVENT 0x01
+#define DETECTION_COMPLETE_EVENT 0x02
+
+/* TS3A227E_REG_KP_INTERRUPT 0x02 */
+#define PRESS_MASK(idx) (0x01 << (2 * (idx)))
+#define RELEASE_MASK(idx) (0x02 << (2 * (idx)))
+
+/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */
+#define INS_REM_INT_DISABLE 0x01
+#define DETECTION_COMPLETE_INT_DISABLE 0x02
+#define ADC_COMPLETE_INT_DISABLE 0x04
+#define INTB_DISABLE 0x08
+
+/* TS3A227E_REG_SETTING_2 0x05 */
+#define KP_ENABLE 0x04
+
+/* TS3A227E_REG_ACCESSORY_STATUS  0x0b */
+#define TYPE_3_POLE 0x01
+#define TYPE_4_POLE_OMTP 0x02
+#define TYPE_4_POLE_STANDARD 0x04
+#define JACK_INSERTED 0x08
+#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD)
+
+static const struct reg_default ts3a227e_reg_defaults[] = {
+	{ TS3A227E_REG_DEVICE_ID, 0x10 },
+	{ TS3A227E_REG_INTERRUPT, 0x00 },
+	{ TS3A227E_REG_KP_INTERRUPT, 0x00 },
+	{ TS3A227E_REG_INTERRUPT_DISABLE, 0x08 },
+	{ TS3A227E_REG_SETTING_1, 0x23 },
+	{ TS3A227E_REG_SETTING_2, 0x00 },
+	{ TS3A227E_REG_SETTING_3, 0x0e },
+	{ TS3A227E_REG_SWITCH_CONTROL_1, 0x00 },
+	{ TS3A227E_REG_SWITCH_CONTROL_2, 0x00 },
+	{ TS3A227E_REG_SWITCH_STATUS_1, 0x0c },
+	{ TS3A227E_REG_SWITCH_STATUS_2, 0x00 },
+	{ TS3A227E_REG_ACCESSORY_STATUS, 0x00 },
+	{ TS3A227E_REG_ADC_OUTPUT, 0x00 },
+	{ TS3A227E_REG_KP_THRESHOLD_1, 0x20 },
+	{ TS3A227E_REG_KP_THRESHOLD_2, 0x40 },
+	{ TS3A227E_REG_KP_THRESHOLD_3, 0x68 },
+};
+
+static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2:
+	case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE:
+	case TS3A227E_REG_SETTING_2:
+	case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static void ts3a227e_jack_report(struct ts3a227e *ts3a227e)
+{
+	unsigned int i;
+	int report = 0;
+
+	if (!ts3a227e->jack)
+		return;
+
+	if (ts3a227e->plugged)
+		report = SND_JACK_HEADPHONE;
+	if (ts3a227e->mic_present)
+		report |= SND_JACK_MICROPHONE;
+	for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) {
+		if (ts3a227e->buttons_held & (1 << i))
+			report |= ts3a227e_buttons[i];
+	}
+	snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK);
+}
+
+static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg)
+{
+	bool plugged, mic_present;
+
+	plugged = !!(acc_reg & JACK_INSERTED);
+	mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK);
+
+	ts3a227e->plugged = plugged;
+
+	if (mic_present != ts3a227e->mic_present) {
+		ts3a227e->mic_present = mic_present;
+		ts3a227e->buttons_held = 0;
+		if (mic_present) {
+			/* Enable key press detection. */
+			regmap_update_bits(ts3a227e->regmap,
+					   TS3A227E_REG_SETTING_2,
+					   KP_ENABLE, KP_ENABLE);
+		}
+	}
+}
+
+static irqreturn_t ts3a227e_interrupt(int irq, void *data)
+{
+	struct ts3a227e *ts3a227e = (struct ts3a227e *)data;
+	struct regmap *regmap = ts3a227e->regmap;
+	unsigned int int_reg, kp_int_reg, acc_reg, i;
+
+	/* Check for plug/unplug. */
+	regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg);
+	if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) {
+		regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg);
+		ts3a227e_new_jack_state(ts3a227e, acc_reg);
+	}
+
+	/* Report any key events. */
+	regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg);
+	for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) {
+		if (kp_int_reg & PRESS_MASK(i))
+			ts3a227e->buttons_held |= (1 << i);
+		if (kp_int_reg & RELEASE_MASK(i))
+			ts3a227e->buttons_held &= ~(1 << i);
+	}
+
+	ts3a227e_jack_report(ts3a227e);
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * ts3a227e_enable_jack_detect - Specify a jack for event reporting
+ *
+ * @component:  component to register the jack with
+ * @jack: jack to use to report headset and button events on
+ *
+ * After this function has been called the headset insert/remove and button
+ * events 0-3 will be routed to the given jack.  Jack can be null to stop
+ * reporting.
+ */
+int ts3a227e_enable_jack_detect(struct snd_soc_component *component,
+				struct snd_soc_jack *jack)
+{
+	struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component);
+
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+	ts3a227e->jack = jack;
+	ts3a227e_jack_report(ts3a227e);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect);
+
+static struct snd_soc_component_driver ts3a227e_soc_driver;
+
+static const struct regmap_config ts3a227e_regmap_config = {
+	.val_bits = 8,
+	.reg_bits = 8,
+
+	.max_register = TS3A227E_REG_KP_THRESHOLD_3,
+	.readable_reg = ts3a227e_readable_reg,
+	.writeable_reg = ts3a227e_writeable_reg,
+	.volatile_reg = ts3a227e_volatile_reg,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = ts3a227e_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults),
+};
+
+static int ts3a227e_i2c_probe(struct i2c_client *i2c,
+			      const struct i2c_device_id *id)
+{
+	struct ts3a227e *ts3a227e;
+	struct device *dev = &i2c->dev;
+	int ret;
+
+	ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL);
+	if (ts3a227e == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, ts3a227e);
+
+	ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config);
+	if (IS_ERR(ts3a227e->regmap))
+		return PTR_ERR(ts3a227e->regmap);
+
+	ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt,
+					IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+					"TS3A227E", ts3a227e);
+	if (ret) {
+		dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_component(&i2c->dev, &ts3a227e_soc_driver,
+					      NULL, 0);
+	if (ret)
+		return ret;
+
+	/* Enable interrupts except for ADC complete. */
+	regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE,
+			   INTB_DISABLE | ADC_COMPLETE_INT_DISABLE,
+			   ADC_COMPLETE_INT_DISABLE);
+
+	return 0;
+}
+
+static const struct i2c_device_id ts3a227e_i2c_ids[] = {
+	{ "ts3a227e", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids);
+
+static const struct of_device_id ts3a227e_of_match[] = {
+	{ .compatible = "ti,ts3a227e", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, ts3a227e_of_match);
+
+static struct i2c_driver ts3a227e_driver = {
+	.driver = {
+		.name = "ts3a227e",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(ts3a227e_of_match),
+	},
+	.probe = ts3a227e_i2c_probe,
+	.id_table = ts3a227e_i2c_ids,
+};
+module_i2c_driver(ts3a227e_driver);
+
+MODULE_DESCRIPTION("ASoC ts3a227e driver");
+MODULE_AUTHOR("Dylan Reid <dgreid@chromium.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h
new file mode 100644
index 000000000000..e2acf9c5bebe
--- /dev/null
+++ b/sound/soc/codecs/ts3a227e.h
@@ -0,0 +1,17 @@
+/*
+ * TS3A227E Autonous Audio Accessory Detection and Configureation Switch
+ *
+ * Copyright (C) 2014 Google, Inc.
+ *
+ * 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 _TS3A227E_H
+#define _TS3A227E_H
+
+int ts3a227e_enable_jack_detect(struct snd_soc_component *component,
+				struct snd_soc_jack *jack);
+
+#endif
-- 
cgit v1.2.3-58-ga151


From f10c0a71e6efc7c8cbc3bfcfd0ecf822607f0b3d Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 26 Nov 2014 20:57:56 +0100
Subject: ASoC: tlv320aic32x4: Cleanup manual bias level transitions

Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.

The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic32x4.c | 24 +-----------------------
 1 file changed, 1 insertion(+), 23 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 6ea662db2410..015467ed606b 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -597,18 +597,6 @@ static struct snd_soc_dai_driver aic32x4_dai = {
 	.symmetric_rates = 1,
 };
 
-static int aic32x4_suspend(struct snd_soc_codec *codec)
-{
-	aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int aic32x4_resume(struct snd_soc_codec *codec)
-{
-	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-
 static int aic32x4_probe(struct snd_soc_codec *codec)
 {
 	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
@@ -654,8 +642,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 		snd_soc_write(codec, AIC32X4_RMICPGANIN,
 				AIC32X4_RMICPGANIN_CM1R_10K);
 
-	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	/*
 	 * Workaround: for an unknown reason, the ADC needs to be powered up
 	 * and down for the first capture to work properly. It seems related to
@@ -669,18 +655,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int aic32x4_remove(struct snd_soc_codec *codec)
-{
-	aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 	.probe = aic32x4_probe,
-	.remove = aic32x4_remove,
-	.suspend = aic32x4_suspend,
-	.resume = aic32x4_resume,
 	.set_bias_level = aic32x4_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = aic32x4_snd_controls,
 	.num_controls = ARRAY_SIZE(aic32x4_snd_controls),
-- 
cgit v1.2.3-58-ga151


From 68f438378cde79e29f71c7e043b10d76001d8892 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 26 Nov 2014 20:57:57 +0100
Subject: ASoC: tlv320aic3x: Cleanup manual bias level transitions

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic3x.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index f0a828119aba..b7ebce054b4e 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1491,7 +1491,6 @@ static int aic3x_remove(struct snd_soc_codec *codec)
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	int i;
 
-	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	list_del(&aic3x->list);
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
 		regulator_unregister_notifier(aic3x->supplies[i].consumer,
-- 
cgit v1.2.3-58-ga151


From 90db15e17e46e2841843bf20f258fed963228bed Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 26 Nov 2014 20:57:58 +0100
Subject: ASoC: tlv320dac33: Cleanup manual bias level transitions

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320dac33.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index e21ed934bdbf..0fe2ced5b09f 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1436,8 +1436,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec)
 {
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
-	dac33_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
 	if (dac33->irq >= 0) {
 		free_irq(dac33->irq, dac33->codec);
 		destroy_workqueue(dac33->dac33_wq);
-- 
cgit v1.2.3-58-ga151


From 3ec8d2036464d961a6314281ae68d80a5c071d07 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 26 Nov 2014 20:57:59 +0100
Subject: ASoC: twl4030: Cleanup manual bias level transitions

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/twl4030.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index b6b0cb399599..27f3b21effb2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -2177,8 +2177,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec)
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	struct twl4030_codec_data *pdata = twl4030->pdata;
 
-	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
 	if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio))
 		gpio_free(pdata->hs_extmute_gpio);
 
-- 
cgit v1.2.3-58-ga151