From 040e3360af3736348112d29425bf5d0be5b93115 Mon Sep 17 00:00:00 2001
From: Francesco Dolcini
Date: Fri, 24 Jun 2022 12:13:01 +0200
Subject: ASoC: sgtl5000: Fix noise on shutdown/remove

Put the SGTL5000 in a silent/safe state on shutdown/remove, this is
required since the SGTL5000 produces a constant noise on its output
after it is configured and its clock is removed. Without this change
this is happening every time the module is unbound/removed or from
reboot till the clock is enabled again.

The issue was experienced on both a Toradex Colibri/Apalis iMX6, but can
be easily reproduced everywhere just playing something on the codec and
after that removing/unbinding the driver.

Fixes: 9b34e6cc3bc2 ("ASoC: Add Freescale SGTL5000 codec support")
Signed-off-by: Francesco Dolcini <francesco.dolcini@toradex.com>
Reviewed-by: Fabio Estevam <festevam@denx.de>
Link: https://lore.kernel.org/r/20220624101301.441314-1-francesco.dolcini@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 9 +++++++++
 sound/soc/codecs/sgtl5000.h | 1 +
 2 files changed, 10 insertions(+)

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 2aa48aef6a97..3363d1696ad7 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1795,6 +1795,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client)
 {
 	struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client);
 
+	regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT);
+	regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT);
+
 	clk_disable_unprepare(sgtl5000->mclk);
 	regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies);
 	regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies);
@@ -1802,6 +1805,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+static void sgtl5000_i2c_shutdown(struct i2c_client *client)
+{
+	sgtl5000_i2c_remove(client);
+}
+
 static const struct i2c_device_id sgtl5000_id[] = {
 	{"sgtl5000", 0},
 	{},
@@ -1822,6 +1830,7 @@ static struct i2c_driver sgtl5000_i2c_driver = {
 	},
 	.probe_new = sgtl5000_i2c_probe,
 	.remove = sgtl5000_i2c_remove,
+	.shutdown = sgtl5000_i2c_shutdown,
 	.id_table = sgtl5000_id,
 };
 
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index 56ec5863f250..3a808c762299 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -80,6 +80,7 @@
 /*
  * SGTL5000_CHIP_DIG_POWER
  */
+#define SGTL5000_DIG_POWER_DEFAULT		0x0000
 #define SGTL5000_ADC_EN				0x0040
 #define SGTL5000_DAC_EN				0x0020
 #define SGTL5000_DAP_POWERUP			0x0010
-- 
cgit v1.2.3


From 242a88ff27b23cbf626f9764955d91200f12b53a Mon Sep 17 00:00:00 2001
From: Sascha Hauer
Date: Fri, 24 Jun 2022 11:26:01 +0200
Subject: ASoC: audio_graph_card2: Fix port numbers in example

The example in audio-graph-card2.c has multiple nodes with the same name
in it. Change the port numbers to get different names.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Link: https://lore.kernel.org/r/20220624092601.2445224-1-s.hauer@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card2.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 77ac4051b827..d34b29a49268 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -90,12 +90,12 @@ links indicates connection part of CPU side (= A).
 			ports@0 {
 (X) (A)			mcpu:	port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; };
 (y)				port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; };
-(y)				port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
+(y)				port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
 			};
 			ports@1 {
 (X)				port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; };
-(y)				port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
-(y)				port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
+(y)				port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
+(y)				port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
 			};
 		};
 	};
-- 
cgit v1.2.3


From f0d96937d31c4615a6418e4bed5cee50a952040e Mon Sep 17 00:00:00 2001
From: David Owens
Date: Mon, 20 Jun 2022 13:37:43 -0500
Subject: ASoC: ti: omap-mcbsp: duplicate sysfs error

Convert to managed versions of sysfs and clk allocation to simplify
unbinding and error handling in probe.  Managed sysfs node
creation specifically addresses the following error seen the second time
probe is attempted after sdma_pcm_platform_register() previously requsted
probe deferral:

sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres'

Signed-off-by: David Owens <dowens@precisionplanting.com>
Link: https://lore.kernel.org/r/20220620183744.3176557-1-dowens@precisionplanting.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/omap-mcbsp-priv.h |  2 --
 sound/soc/ti/omap-mcbsp-st.c   | 14 ++------------
 sound/soc/ti/omap-mcbsp.c      | 19 ++-----------------
 3 files changed, 4 insertions(+), 31 deletions(-)

diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h
index 7865cda4bf0a..da519ea1f303 100644
--- a/sound/soc/ti/omap-mcbsp-priv.h
+++ b/sound/soc/ti/omap-mcbsp-priv.h
@@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg,
 
 /* Sidetone specific API */
 int omap_mcbsp_st_init(struct platform_device *pdev);
-void omap_mcbsp_st_cleanup(struct platform_device *pdev);
-
 int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp);
 int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp);
 
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 0bc7d26c660a..7e8179cae92e 100644
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
 	if (!st_data)
 		return -ENOMEM;
 
-	st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick");
+	st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick");
 	if (IS_ERR(st_data->mcbsp_iclk)) {
 		dev_warn(mcbsp->dev,
 			 "Failed to get ick, sidetone might be broken\n");
@@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
 	if (!st_data->io_base_st)
 		return -ENOMEM;
 
-	ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group);
+	ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group);
 	if (ret)
 		return ret;
 
@@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
 	return 0;
 }
 
-void omap_mcbsp_st_cleanup(struct platform_device *pdev)
-{
-	struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
-
-	if (mcbsp->st_data) {
-		sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group);
-		clk_put(mcbsp->st_data->mcbsp_iclk);
-	}
-}
-
 static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
 				    struct snd_ctl_elem_info *uinfo)
 {
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 4479d74f0a45..9933b33c80ca 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 		mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10;
 		mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10;
 
-		ret = sysfs_create_group(&mcbsp->dev->kobj,
-					 &additional_attr_group);
+		ret = devm_device_add_group(mcbsp->dev, &additional_attr_group);
 		if (ret) {
 			dev_err(mcbsp->dev,
 				"Unable to create additional controls\n");
@@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 		}
 	}
 
-	ret = omap_mcbsp_st_init(pdev);
-	if (ret)
-		goto err_st;
-
-	return 0;
-
-err_st:
-	if (mcbsp->pdata->buffer_size)
-		sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-	return ret;
+	return omap_mcbsp_st_init(pdev);
 }
 
 /*
@@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
 	if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req))
 		cpu_latency_qos_remove_request(&mcbsp->pm_qos_req);
 
-	if (mcbsp->pdata->buffer_size)
-		sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-
-	omap_mcbsp_st_cleanup(pdev);
-
 	return 0;
 }
 
-- 
cgit v1.2.3


From 9e51ac6ea6041641f64fb761ccf9de271e6576cb Mon Sep 17 00:00:00 2001
From: Daniel Mack
Date: Fri, 24 Jun 2022 12:47:10 +0200
Subject: ASoC: max98396: Fix register access for PCM format settings

max98396_dai_set_fmt() modifes register 2041 and touches bits in the mask
0x3a. Make sure to use the right mask for that operation.

Signed-off-by: Daniel Mack <daniel@zonque.org>
Link: https://lore.kernel.org/r/20220624104712.1934484-7-daniel@zonque.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98396.c | 10 ++++++----
 1 file changed, 6 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c
index 56eb62bb041f..34db38812807 100644
--- a/sound/soc/codecs/max98396.c
+++ b/sound/soc/codecs/max98396.c
@@ -342,12 +342,15 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component);
-	unsigned int format = 0;
+	unsigned int format_mask, format = 0;
 	unsigned int bclk_pol = 0;
 	int ret, status;
 	int reg;
 	bool update = false;
 
+	format_mask = MAX98396_PCM_MODE_CFG_FORMAT_MASK |
+		      MAX98396_PCM_MODE_CFG_LRCLKEDGE;
+
 	dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt);
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -395,7 +398,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		ret = regmap_read(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, &reg);
 		if (ret < 0)
 			return -EINVAL;
-		if (format != (reg & MAX98396_PCM_BCLKEDGE_BSEL_MASK)) {
+		if (format != (reg & format_mask)) {
 			update = true;
 		} else {
 			ret = regmap_read(max98396->regmap,
@@ -412,8 +415,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 
 	regmap_update_bits(max98396->regmap,
 			   MAX98396_R2041_PCM_MODE_CFG,
-			   MAX98396_PCM_BCLKEDGE_BSEL_MASK,
-			   format);
+			   format_mask, format);
 
 	regmap_update_bits(max98396->regmap,
 			   MAX98396_R2042_PCM_CLK_SETUP,
-- 
cgit v1.2.3


From 7d90c8e6396ba245da16bedd789df6d669375408 Mon Sep 17 00:00:00 2001
From: Sascha Hauer
Date: Fri, 24 Jun 2022 12:57:16 +0200
Subject: ASoC: tlv320adcx140: Fix tx_mask check

The tx_mask check doesn't reflect what the driver and the chip support.

The check currently checks for exactly two slots being enabled. The
tlv320adcx140 supports anything between one and eight channels, so relax
the check accordingly.

The tlv320adcx140 supports arbitrary tx_mask settings, but the driver
currently only supports adjacent slots beginning with the first slot,
so extend the check to check that the first slot is being used and that
there are no holes in the tx_mask.

Leave a comment to make it's the driver that limits the tx_mask
settings, not the chip itself.

While at it remove the set-but-unused struct adcx140p_priv::tdm_delay
field.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Link: https://lore.kernel.org/r/20220624105716.2579539-1-s.hauer@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320adcx140.c | 13 ++++++-------
 1 file changed, 6 insertions(+), 7 deletions(-)

diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
index b55f0b836932..0b729658fde8 100644
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -33,7 +33,6 @@ struct adcx140_priv {
 	bool micbias_vg;
 
 	unsigned int dai_fmt;
-	unsigned int tdm_delay;
 	unsigned int slot_width;
 };
 
@@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_component *component = codec_dai->component;
 	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
-	unsigned int lsb;
 
-	/* TDM based on DSP mode requires slots to be adjacent */
-	lsb = __ffs(tx_mask);
-	if ((lsb + 1) != __fls(tx_mask)) {
-		dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
+	/*
+	 * The chip itself supports arbitrary masks, but the driver currently
+	 * only supports adjacent slots beginning at the first slot.
+	 */
+	if (tx_mask != GENMASK(__fls(tx_mask), 0)) {
+		dev_err(component->dev, "Only lower adjacent slots are supported\n");
 		return -EINVAL;
 	}
 
@@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	adcx140->tdm_delay = lsb;
 	adcx140->slot_width = slot_width;
 
 	return 0;
-- 
cgit v1.2.3


From a5d6d28e2ea38dff017cb562dfbe0259d093a851 Mon Sep 17 00:00:00 2001
From: Yassine Oudjana
Date: Wed, 22 Jun 2022 10:17:45 +0400
Subject: ASoC: wcd9335: Use int array instead of bitmask for TX mixers

Currently slim_tx_mixer_get reports all TX mixers as enabled when
at least one is, due to it reading the entire tx_port_value bitmask
without testing the specific bit corresponding to a TX port.
Furthermore, using the same bitmask for all capture DAIs makes
setting one mixer affect them all. To prevent this, and since
the SLIM TX muxes effectively only connect to one of the mixers
at a time, turn tx_port_value into an int array storing the DAI
index each of the ports is connected to.

Signed-off-by: Yassine Oudjana <y.oudjana@protonmail.com>
Link: https://lore.kernel.org/r/20220622061745.35399-1-y.oudjana@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd9335.c | 17 +++++++++++------
 1 file changed, 11 insertions(+), 6 deletions(-)

diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index d9f135200688..3cb7a3eab8c7 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -342,7 +342,7 @@ struct wcd9335_codec {
 	struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY];
 
 	unsigned int rx_port_value[WCD9335_RX_MAX];
-	unsigned int tx_port_value;
+	unsigned int tx_port_value[WCD9335_TX_MAX];
 	int hph_l_gain;
 	int hph_r_gain;
 	u32 rx_bias_count;
@@ -1334,8 +1334,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc,
 
 	struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc);
 	struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev);
+	struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc);
+	struct soc_mixer_control *mixer =
+			(struct soc_mixer_control *)kc->private_value;
+	int dai_id = widget->shift;
+	int port_id = mixer->shift;
 
-	ucontrol->value.integer.value[0] = wcd->tx_port_value;
+	ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id;
 
 	return 0;
 }
@@ -1358,12 +1363,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc,
 	case AIF2_CAP:
 	case AIF3_CAP:
 		/* only add to the list if value not set */
-		if (enable && !(wcd->tx_port_value & BIT(port_id))) {
-			wcd->tx_port_value |= BIT(port_id);
+		if (enable && wcd->tx_port_value[port_id] != dai_id) {
+			wcd->tx_port_value[port_id] = dai_id;
 			list_add_tail(&wcd->tx_chs[port_id].list,
 					&wcd->dai[dai_id].slim_ch_list);
-		} else if (!enable && (wcd->tx_port_value & BIT(port_id))) {
-			wcd->tx_port_value &= ~BIT(port_id);
+		} else if (!enable && wcd->tx_port_value[port_id] == dai_id) {
+			wcd->tx_port_value[port_id] = -1;
 			list_del_init(&wcd->tx_chs[port_id].list);
 		}
 		break;
-- 
cgit v1.2.3


From 71b5ab96ffe6589abe7a2e302b83f7a426ebe099 Mon Sep 17 00:00:00 2001
From: Charles Keepax
Date: Tue, 28 Jun 2022 16:34:06 +0100
Subject: ASoC: wm5102: Fix event generation for output compensation

The output compensation controls always returns zero regardless of if
the control value was updated. This results in missing notifications
to user-space of the control change. Update the handling to return 1
when the value is changed.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220628153409.3266932-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm5102.c | 21 +++++++++++++++++----
 1 file changed, 17 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index da2f8998df87..b034df47a5ef 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+	uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+	int ret = 0;
 
 	mutex_lock(&arizona->dac_comp_lock);
-	arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+	if (arizona->dac_comp_coeff != dac_comp_coeff) {
+		arizona->dac_comp_coeff = dac_comp_coeff;
+		ret = 1;
+	}
 	mutex_unlock(&arizona->dac_comp_lock);
 
-	return 0;
+	return ret;
 }
 
 static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol,
@@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+	struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+	int ret = 0;
+
+	if (ucontrol->value.integer.value[0] > mc->max)
+		return -EINVAL;
 
 	mutex_lock(&arizona->dac_comp_lock);
-	arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+	if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) {
+		arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+		ret = 1;
+	}
 	mutex_unlock(&arizona->dac_comp_lock);
 
-	return 0;
+	return ret;
 }
 
 static const char * const wm5102_osr_text[] = {
-- 
cgit v1.2.3


From 15b2e5d10ccf32a1a1ae7c636511e2f51320fdb5 Mon Sep 17 00:00:00 2001
From: Charles Keepax
Date: Tue, 28 Jun 2022 16:34:07 +0100
Subject: ASoC: wm8998: Fix event generation for input mux

wm8998_inmux_put returns the value of snd_soc_dapm_mux_update_power,
which returns a 1 if a path was found for the kcontrol. This is
obviously different to the expected return a 1 if the control
was updated value. This results in spurious notifications to
user-space. Update the handling to only return a 1 when the value is
changed.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220628153409.3266932-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8998.c | 21 ++++++++++++++-------
 1 file changed, 14 insertions(+), 7 deletions(-)

diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 00b59fc9b1fe..ab5481187c71 100644
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int mode_reg, mode_index;
 	unsigned int mux, inmode, src_val, mode_val;
+	int change, ret;
 
 	mux = ucontrol->value.enumerated.item[0];
 	if (mux > 1)
@@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
 	snd_soc_component_update_bits(component, mode_reg,
 				      ARIZONA_IN1_MODE_MASK, mode_val);
 
-	snd_soc_component_update_bits(component, e->reg,
-				      ARIZONA_IN1L_SRC_MASK |
-				      ARIZONA_IN1L_SRC_SE_MASK,
-				      src_val);
+	change = snd_soc_component_update_bits(component, e->reg,
+					       ARIZONA_IN1L_SRC_MASK |
+					       ARIZONA_IN1L_SRC_SE_MASK,
+					       src_val);
 
-	return snd_soc_dapm_mux_update_power(dapm, kcontrol,
-					     ucontrol->value.enumerated.item[0],
-					     e, NULL);
+	ret = snd_soc_dapm_mux_update_power(dapm, kcontrol,
+					    ucontrol->value.enumerated.item[0],
+					    e, NULL);
+	if (ret < 0) {
+		dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret);
+		return ret;
+	}
+
+	return change;
 }
 
 static const char * const wm8998_inmux_texts[] = {
-- 
cgit v1.2.3


From 870d72ab9228575b2f005c9a23ea08787e0f63e6 Mon Sep 17 00:00:00 2001
From: Charles Keepax
Date: Tue, 28 Jun 2022 16:34:08 +0100
Subject: ASoC: cs47l92: Fix event generation for OUT1 demux

cs47l92_put_demux returns the value of snd_soc_dapm_mux_update_power,
which returns a 1 if a path was found for the kcontrol. This is
obviously different to the expected return a 1 if the control
was updated value. This results in spurious notifications to
user-space. Update the handling to only return a 1 when the value is
changed.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220628153409.3266932-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs47l92.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c
index a1b8dcdb9f7b..444026b7d54b 100644
--- a/sound/soc/codecs/cs47l92.c
+++ b/sound/soc/codecs/cs47l92.c
@@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol,
 end:
 	snd_soc_dapm_mutex_unlock(dapm);
 
-	return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+	ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+	if (ret < 0) {
+		dev_err(madera->dev, "Failed to update demux power state: %d\n", ret);
+		return ret;
+	}
+
+	return change;
 }
 
 static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum,
-- 
cgit v1.2.3


From f99e930655f411453170a5f332e12c2d2748822e Mon Sep 17 00:00:00 2001
From: Charles Keepax
Date: Tue, 28 Jun 2022 16:34:09 +0100
Subject: ASoC: arizona: Update arizona_aif_cfg_changed to use RX_BCLK_RATE

Currently the function arizona_aif_cfg_changed uses the TX_BCLK_RATE,
however this register is not used on wm8998. This was not noticed as
previously snd_soc_component_read did not print an error message.
However, now the log gets filled with error messages, further more the
test for if the LRCLK changed will return spurious results.

Update the code to use the RX_BCLK_RATE register, the LRCLK parameters
are written to both registers and the RX_BCLK_RATE register is used
across all Arizona devices.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220628153409.3266932-4-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(-)

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e32871b3f68a..7434aeeda292 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component,
 	if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
 		return true;
 
-	val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE);
-	if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
+	val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE);
+	if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK))
 		return true;
 
 	val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1);
-- 
cgit v1.2.3


From 0735e400fa07a3687797f241f5b74d77a6d383c7 Mon Sep 17 00:00:00 2001
From: Marek Vasut
Date: Tue, 28 Jun 2022 18:58:40 +0200
Subject: ASoC: doc: Capitalize RESET line name

Make sure all AC97 interface lines are spelled in capitals,
to avoid confusing readers about where the 5th line is.

Signed-off-by: Marek Vasut <marex@denx.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20220628165840.152235-1-marex@denx.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/sound/soc/dai.rst | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst
index 009b07e5a0f3..bf8431386d26 100644
--- a/Documentation/sound/soc/dai.rst
+++ b/Documentation/sound/soc/dai.rst
@@ -10,7 +10,7 @@ AC97
 ====
 
 AC97 is a five wire interface commonly found on many PC sound cards. It is
-now also popular in many portable devices. This DAI has a reset line and time
+now also popular in many portable devices. This DAI has a RESET line and time
 multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
 The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
 frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
-- 
cgit v1.2.3


From 94c65dffd4c4af052b3ea8934fbcb2fa8da276a8 Mon Sep 17 00:00:00 2001
From: Bryan O'Donoghue
Date: Wed, 29 Jun 2022 12:40:12 +0100
Subject: ASoC: dt-bindings: Fix description for msm8916

For the existing msm8916 bindings the minimum reg/reg-names is 1 not 2.
Similarly the minimum interrupt/interrupt-names is 1 not 2.

Fixes: f3fc4fbfa2d2 ("ASoC: dt-bindings: Add SC7280 lpass cpu bindings")
Signed-off-by: Bryan O'Donoghue <bryan.odonoghue@linaro.org>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20220629114012.3282945-1-bryan.odonoghue@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
index e9a533080b32..ef18a572a1ff 100644
--- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
@@ -25,12 +25,12 @@ properties:
       - qcom,sc7280-lpass-cpu
 
   reg:
-    minItems: 2
+    minItems: 1
     maxItems: 6
     description: LPAIF core registers
 
   reg-names:
-    minItems: 2
+    minItems: 1
     maxItems: 6
 
   clocks:
@@ -42,12 +42,12 @@ properties:
     maxItems: 10
 
   interrupts:
-    minItems: 2
+    minItems: 1
     maxItems: 4
     description: LPAIF DMA buffer interrupt
 
   interrupt-names:
-    minItems: 2
+    minItems: 1
     maxItems: 4
 
   qcom,adsp:
-- 
cgit v1.2.3


From cd10bb89b0d57bca98eb75e0444854a1c129a14e Mon Sep 17 00:00:00 2001
From: Martin Povišer
Date: Thu, 30 Jun 2022 09:51:32 +0200
Subject: ASoC: tas2764: Add post reset delays

Make sure there is at least 1 ms delay from reset to first command as
is specified in the datasheet. This is a fix similar to commit
307f31452078 ("ASoC: tas2770: Insert post reset delay").

Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-1-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2764.c | 10 ++++++++--
 1 file changed, 8 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index d395feffb30b..d702a39eaa7b 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764)
 		gpiod_set_value_cansleep(tas2764->reset_gpio, 0);
 		msleep(20);
 		gpiod_set_value_cansleep(tas2764->reset_gpio, 1);
+		usleep_range(1000, 2000);
 	}
 
 	snd_soc_component_write(tas2764->component, TAS2764_SW_RST,
 				TAS2764_RST);
+	usleep_range(1000, 2000);
 }
 
 static int tas2764_set_bias_level(struct snd_soc_component *component,
@@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component)
 	struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
 	int ret;
 
-	if (tas2764->sdz_gpio)
+	if (tas2764->sdz_gpio) {
 		gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+		usleep_range(1000, 2000);
+	}
 
 	ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL,
 					    TAS2764_PWR_CTRL_MASK,
@@ -501,8 +505,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
 
 	tas2764->component = component;
 
-	if (tas2764->sdz_gpio)
+	if (tas2764->sdz_gpio) {
 		gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+		usleep_range(1000, 2000);
+	}
 
 	tas2764_reset(tas2764);
 
-- 
cgit v1.2.3


From d1a10f1b48202e2d183cce144c218a211e98d906 Mon Sep 17 00:00:00 2001
From: Martin Povišer
Date: Thu, 30 Jun 2022 09:51:33 +0200
Subject: ASoC: tas2764: Fix and extend FSYNC polarity handling

Fix setting of FSYNC polarity in case of LEFT_J and DSP_A/B formats.
Do NOT set the SCFG field as was previously done, because that is not
correct and is also in conflict with the "ASI1 Source" control which
sets the same SCFG field!

Also add support for explicit polarity inversion.

Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-2-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2764.c | 30 +++++++++++++++++-------------
 sound/soc/codecs/tas2764.h |  6 ++----
 2 files changed, 19 insertions(+), 17 deletions(-)

diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index d702a39eaa7b..2d045631bfe9 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -135,7 +135,8 @@ static const char * const tas2764_ASI1_src[] = {
 };
 
 static SOC_ENUM_SINGLE_DECL(
-	tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, 4, tas2764_ASI1_src);
+	tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, TAS2764_TDM_CFG2_SCFG_SHIFT,
+	tas2764_ASI1_src);
 
 static const struct snd_kcontrol_new tas2764_asi1_mux =
 	SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum);
@@ -333,20 +334,22 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
-	u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0;
-	int iface;
+	u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
 	int ret;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_IF:
+		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+		fallthrough;
 	case SND_SOC_DAIFMT_NB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
 		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+		fallthrough;
 	case SND_SOC_DAIFMT_IB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
 		break;
-	default:
-		dev_err(tas2764->dev, "ASI format Inverse is not found\n");
-		return -EINVAL;
 	}
 
 	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
@@ -357,13 +360,13 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
+		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+		fallthrough;
 	case SND_SOC_DAIFMT_DSP_A:
-		iface = TAS2764_TDM_CFG2_SCFG_I2S;
 		tdm_rx_start_slot = 1;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 	case SND_SOC_DAIFMT_LEFT_J:
-		iface = TAS2764_TDM_CFG2_SCFG_LEFT_J;
 		tdm_rx_start_slot = 0;
 		break;
 	default:
@@ -372,14 +375,15 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
-	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
-					    TAS2764_TDM_CFG1_MASK,
-					    (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
+	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG0,
+					    TAS2764_TDM_CFG0_FRAME_START,
+					    asi_cfg_0);
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG2,
-					    TAS2764_TDM_CFG2_SCFG_MASK, iface);
+	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
+					    TAS2764_TDM_CFG1_MASK,
+					    (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h
index 67d6fd903c42..f015f22a083b 100644
--- a/sound/soc/codecs/tas2764.h
+++ b/sound/soc/codecs/tas2764.h
@@ -47,6 +47,7 @@
 #define TAS2764_TDM_CFG0_MASK		GENMASK(3, 1)
 #define TAS2764_TDM_CFG0_44_1_48KHZ	BIT(3)
 #define TAS2764_TDM_CFG0_88_2_96KHZ	(BIT(3) | BIT(1))
+#define TAS2764_TDM_CFG0_FRAME_START	BIT(0)
 
 /* TDM Configuration Reg1 */
 #define TAS2764_TDM_CFG1		TAS2764_REG(0X0, 0x09)
@@ -66,10 +67,7 @@
 #define TAS2764_TDM_CFG2_RXS_16BITS	0x0
 #define TAS2764_TDM_CFG2_RXS_24BITS	BIT(0)
 #define TAS2764_TDM_CFG2_RXS_32BITS	BIT(1)
-#define TAS2764_TDM_CFG2_SCFG_MASK	GENMASK(5, 4)
-#define TAS2764_TDM_CFG2_SCFG_I2S	0x0
-#define TAS2764_TDM_CFG2_SCFG_LEFT_J	BIT(4)
-#define TAS2764_TDM_CFG2_SCFG_RIGHT_J	BIT(5)
+#define TAS2764_TDM_CFG2_SCFG_SHIFT	4
 
 /* TDM Configuration Reg3 */
 #define TAS2764_TDM_CFG3		TAS2764_REG(0X0, 0x0c)
-- 
cgit v1.2.3


From 3e99e5697e1f7120b5abc755e8a560b22612d6ed Mon Sep 17 00:00:00 2001
From: Hector Martin
Date: Thu, 30 Jun 2022 09:51:34 +0200
Subject: ASoC: tas2764: Correct playback volume range

DVC value 0xc8 is -100dB and 0xc9 is mute; this needs to map to
-100.5dB as far as the dB scale is concerned. Fix that and enable
the mute flag, so alsamixer correctly shows the control as
<0 dB .. -100 dB, mute>.

Signed-off-by: Hector Martin <marcan@marcan.st>
Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-3-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2764.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index 2d045631bfe9..2bcb407c3cdd 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -536,7 +536,7 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
 }
 
 static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0);
-static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10000, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1);
 
 static const struct snd_kcontrol_new tas2764_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0,
-- 
cgit v1.2.3


From 1c4f29ec878bbf1cc0a1eb54ae7da5ff98e19641 Mon Sep 17 00:00:00 2001
From: Hector Martin
Date: Thu, 30 Jun 2022 09:51:35 +0200
Subject: ASoC: tas2764: Fix amp gain register offset & default

The register default is 0x28 per the datasheet, and the amp gain field
is supposed to be shifted left by one. With the wrong default, the ALSA
controls lie about the power-up state. With the wrong shift, we get only
half the gain we expect.

Signed-off-by: Hector Martin <marcan@marcan.st>
Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-4-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2764.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index 2bcb407c3cdd..4cb788f3e5f7 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -541,7 +541,7 @@ static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1);
 static const struct snd_kcontrol_new tas2764_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0,
 		       TAS2764_DVC_MAX, 1, tas2764_playback_volume),
-	SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 0, 0x14, 0,
+	SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0,
 		       tas2764_digital_tlv),
 };
 
@@ -566,7 +566,7 @@ static const struct reg_default tas2764_reg_defaults[] = {
 	{ TAS2764_SW_RST, 0x00 },
 	{ TAS2764_PWR_CTRL, 0x1a },
 	{ TAS2764_DVC, 0x00 },
-	{ TAS2764_CHNL_0, 0x00 },
+	{ TAS2764_CHNL_0, 0x28 },
 	{ TAS2764_TDM_CFG0, 0x09 },
 	{ TAS2764_TDM_CFG1, 0x02 },
 	{ TAS2764_TDM_CFG2, 0x0a },
-- 
cgit v1.2.3


From df5b4aca7248dc5a5cae93f162eae0decf972e48 Mon Sep 17 00:00:00 2001
From: Jianglei Nie
Date: Thu, 30 Jun 2022 02:25:20 +0800
Subject: ASoC: qdsp6: fix potential memory leak in
 q6apm_get_audioreach_graph()

q6apm_get_audioreach_graph() allocates a memory chunk for graph->graph
with audioreach_alloc_graph_pkt(). When idr_alloc() fails, graph->graph
is not released, which will lead to a memory leak.

We can release the graph->graph with kfree() when idr_alloc() fails to
fix the memory leak.

Signed-off-by: Jianglei Nie <niejianglei2021@163.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20220629182520.2164409-1-niejianglei2021@163.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6apm.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index f424d7aa389a..794019286c70 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui
 	id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL);
 	if (id < 0) {
 		dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id);
+		kfree(graph->graph);
 		kfree(graph);
 		mutex_unlock(&apm->lock);
 		return ERR_PTR(id);
-- 
cgit v1.2.3


From c7dab6745f4288af1e45d3809bf86a3778301616 Mon Sep 17 00:00:00 2001
From: Brent Lu
Date: Fri, 1 Jul 2022 22:15:17 +0800
Subject: ASoC: Intel: sof_rt5682: fix out-of-bounds array access

Starting from ADL platform we have four HDMI PCM devices which exceeds
the size of sof_hdmi array. Since each sof_hdmi_pcm structure
represents one HDMI PCM device, we remove the sof_hdmi array and add a
new member hdmi_jack to the sof_hdmi_pcm structure to fix the
out-of-bounds problem.

Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20220701141517.264070-1-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/sof_rt5682.c | 10 +++-------
 1 file changed, 3 insertions(+), 7 deletions(-)

diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 5d67a2c87a1d..4a90a0a5d831 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -69,11 +69,10 @@ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
 
 static int is_legacy_cpu;
 
-static struct snd_soc_jack sof_hdmi[3];
-
 struct sof_hdmi_pcm {
 	struct list_head head;
 	struct snd_soc_dai *codec_dai;
+	struct snd_soc_jack hdmi_jack;
 	int device;
 };
 
@@ -434,7 +433,6 @@ static int sof_card_late_probe(struct snd_soc_card *card)
 	char jack_name[NAME_SIZE];
 	struct sof_hdmi_pcm *pcm;
 	int err;
-	int i = 0;
 
 	/* HDMI is not supported by SOF on Baytrail/CherryTrail */
 	if (is_legacy_cpu || !ctx->idisp_codec)
@@ -455,17 +453,15 @@ static int sof_card_late_probe(struct snd_soc_card *card)
 		snprintf(jack_name, sizeof(jack_name),
 			 "HDMI/DP, pcm=%d Jack", pcm->device);
 		err = snd_soc_card_jack_new(card, jack_name,
-					    SND_JACK_AVOUT, &sof_hdmi[i]);
+					    SND_JACK_AVOUT, &pcm->hdmi_jack);
 
 		if (err)
 			return err;
 
 		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
-					  &sof_hdmi[i]);
+					  &pcm->hdmi_jack);
 		if (err < 0)
 			return err;
-
-		i++;
 	}
 
 	if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) {
-- 
cgit v1.2.3


From 051dade346957d5b68ad986f497835805fa7a9dd Mon Sep 17 00:00:00 2001
From: Oder Chiou
Date: Tue, 5 Jul 2022 18:11:33 +0800
Subject: ASoC: rt5640: Fix the wrong state of JD1 and JD2

The patch fixes the wrong state of JD1 and JD2 while the bst1 or bst2 is
power on in the HDA JD using.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Reported-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20220705101134.16792-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5640.c | 30 ++++++++++++++++++++++--------
 1 file changed, 22 insertions(+), 8 deletions(-)

diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 69c80d80ed9d..18b3da9211e3 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component,
 		snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000);
 		snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000);
 		snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000);
-		snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000);
+		if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
+			snd_soc_component_write(component, RT5640_PWR_ANLG1,
+				0x0018);
+		else
+			snd_soc_component_write(component, RT5640_PWR_ANLG1,
+				0x0000);
 		snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000);
 		break;
 
@@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work)
 static irqreturn_t rt5640_irq(int irq, void *data)
 {
 	struct rt5640_priv *rt5640 = data;
+	int delay = 0;
+
+	if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
+		cancel_delayed_work_sync(&rt5640->jack_work);
+		delay = 100;
+	}
 
 	if (rt5640->jack)
-		queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
+		queue_delayed_work(system_long_wq, &rt5640->jack_work, delay);
 
 	return IRQ_HANDLED;
 }
@@ -2580,6 +2591,12 @@ static void rt5640_enable_hda_jack_detect(
 
 	snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0);
 
+	snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+		RT5640_PWR_VREF2, RT5640_PWR_VREF2);
+	usleep_range(10000, 15000);
+	snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV2, RT5640_PWR_FV2);
+
 	rt5640->jack = jack;
 
 	ret = request_irq(rt5640->irq, rt5640_irq,
@@ -2696,16 +2713,13 @@ static int rt5640_probe(struct snd_soc_component *component)
 
 	if (device_property_read_u32(component->dev,
 				     "realtek,jack-detect-source", &val) == 0) {
-		if (val <= RT5640_JD_SRC_GPIO4) {
+		if (val <= RT5640_JD_SRC_GPIO4)
 			rt5640->jd_src = val << RT5640_JD_SFT;
-		} else if (val == RT5640_JD_SRC_HDA_HEADER) {
+		else if (val == RT5640_JD_SRC_HDA_HEADER)
 			rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER;
-			snd_soc_component_update_bits(component, RT5640_DUMMY1,
-				0x0300, 0x0);
-		} else {
+		else
 			dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n",
 				 val);
-		}
 	}
 
 	if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted"))
-- 
cgit v1.2.3


From 219af251bd1694bce1f627d238347d2eaf13de61 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi
Date: Thu, 30 Jun 2022 09:56:37 +0300
Subject: ASoC: Intel: Skylake: Correct the ssp rate discovery in
 skl_get_ssp_clks()

The present flag is only set once when one rate has been found to be saved.
This will effectively going to ignore any rate discovered at later time and
based on the code, this is not the intention.

Fixes: bc2bd45b1f7f3 ("ASoC: Intel: Skylake: Parse nhlt and register clock device")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220630065638.11183-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-nhlt.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 2439a574ac2f..366f7bd9bc02 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -99,7 +99,6 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 	struct nhlt_fmt_cfg *fmt_cfg;
 	struct wav_fmt_ext *wav_fmt;
 	unsigned long rate;
-	bool present = false;
 	int rate_index = 0;
 	u16 channels, bps;
 	u8 clk_src;
@@ -113,6 +112,8 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 		return;
 
 	for (i = 0; i < fmt->fmt_count; i++) {
+		bool present = false;
+
 		fmt_cfg = &fmt->fmt_config[i];
 		wav_fmt = &fmt_cfg->fmt_ext;
 
-- 
cgit v1.2.3


From fc976f5629afb4160ee77798b14a693eac903ffd Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi
Date: Thu, 30 Jun 2022 09:56:38 +0300
Subject: ASoC: Intel: Skylake: Correct the handling of fmt_config flexible
 array

The struct nhlt_format's fmt_config is a flexible array, it must not be
used as normal array.
When moving to the next nhlt_fmt_cfg we need to take into account the data
behind the ->config.caps (indicated by ->config.size).

The logic of the code also changed: it is no longer saves the _last_
fmt_cfg for all found rates.

Fixes: bc2bd45b1f7f3 ("ASoC: Intel: Skylake: Parse nhlt and register clock device")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220630065638.11183-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-nhlt.c | 37 +++++++++++++++++++++++++------------
 1 file changed, 25 insertions(+), 12 deletions(-)

diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 366f7bd9bc02..deb7b820325e 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -111,11 +111,12 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 	if (fmt->fmt_count == 0)
 		return;
 
+	fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
 	for (i = 0; i < fmt->fmt_count; i++) {
+		struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg;
 		bool present = false;
 
-		fmt_cfg = &fmt->fmt_config[i];
-		wav_fmt = &fmt_cfg->fmt_ext;
+		wav_fmt = &saved_fmt_cfg->fmt_ext;
 
 		channels = wav_fmt->fmt.channels;
 		bps = wav_fmt->fmt.bits_per_sample;
@@ -133,12 +134,18 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 		 * derive the rate.
 		 */
 		for (j = i; j < fmt->fmt_count; j++) {
-			fmt_cfg = &fmt->fmt_config[j];
-			wav_fmt = &fmt_cfg->fmt_ext;
+			struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg;
+
+			wav_fmt = &tmp_fmt_cfg->fmt_ext;
 			if ((fs == wav_fmt->fmt.samples_per_sec) &&
-			   (bps == wav_fmt->fmt.bits_per_sample))
+			   (bps == wav_fmt->fmt.bits_per_sample)) {
 				channels = max_t(u16, channels,
 						wav_fmt->fmt.channels);
+				saved_fmt_cfg = tmp_fmt_cfg;
+			}
+			/* Move to the next nhlt_fmt_cfg */
+			tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps +
+							      tmp_fmt_cfg->config.size);
 		}
 
 		rate = channels * bps * fs;
@@ -154,8 +161,11 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 
 		/* Fill rate and parent for sclk/sclkfs */
 		if (!present) {
+			struct nhlt_fmt_cfg *first_fmt_cfg;
+
+			first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
 			i2s_config_ext = (struct skl_i2s_config_blob_ext *)
-						fmt->fmt_config[0].config.caps;
+						first_fmt_cfg->config.caps;
 
 			/* MCLK Divider Source Select */
 			if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -169,6 +179,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 
 			parent = skl_get_parent_clk(clk_src);
 
+			/* Move to the next nhlt_fmt_cfg */
+			fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps +
+							  fmt_cfg->config.size);
 			/*
 			 * Do not copy the config data if there is no parent
 			 * clock available for this clock source select
@@ -177,9 +190,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 				continue;
 
 			sclk[id].rate_cfg[rate_index].rate = rate;
-			sclk[id].rate_cfg[rate_index].config = fmt_cfg;
+			sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg;
 			sclkfs[id].rate_cfg[rate_index].rate = rate;
-			sclkfs[id].rate_cfg[rate_index].config = fmt_cfg;
+			sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg;
 			sclk[id].parent_name = parent->name;
 			sclkfs[id].parent_name = parent->name;
 
@@ -193,13 +206,13 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
 {
 	struct skl_i2s_config_blob_ext *i2s_config_ext;
 	struct skl_i2s_config_blob_legacy *i2s_config;
-	struct nhlt_specific_cfg *fmt_cfg;
+	struct nhlt_fmt_cfg *fmt_cfg;
 	struct skl_clk_parent_src *parent;
 	u32 clkdiv, div_ratio;
 	u8 clk_src;
 
-	fmt_cfg = &fmt->fmt_config[0].config;
-	i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->caps;
+	fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
+	i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps;
 
 	/* MCLK Divider Source Select and divider */
 	if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -228,7 +241,7 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
 		return;
 
 	mclk[id].rate_cfg[0].rate = parent->rate/div_ratio;
-	mclk[id].rate_cfg[0].config = &fmt->fmt_config[0];
+	mclk[id].rate_cfg[0].config = fmt_cfg;
 	mclk[id].parent_name = parent->name;
 }
 
-- 
cgit v1.2.3


From 26b9f2fa7b1c6aba6fa9b83274a3e54868f69562 Mon Sep 17 00:00:00 2001
From: Alexandru Elisei
Date: Mon, 11 Jul 2022 14:05:22 +0100
Subject: ASoC: rockchip: i2s: Fix NULL pointer dereference when pinctrl is not
 found

Commit a5450aba737d ("ASoC: rockchip: i2s: switch BCLK to GPIO") switched
BCLK to GPIO functions when probing the i2s bus interface, but missed
adding a check for when devm_pinctrl_get() returns an error.  This can lead
to the following NULL pointer dereference on a rockpro64-v2 if there are no
"pinctrl" properties in the i2s device tree node.

Check that i2s->pinctrl is valid before attempting to search for the
bclk_on and bclk_off pinctrl states.

Fixes: a5450aba737d ("ASoC: rockchip: i2s: switch BCLK to GPIO")
Signed-off-by: Alexandru Elisei <alexandru.elisei@arm.com>
Link: https://lore.kernel.org/r/20220711130522.401551-1-alexandru.elisei@arm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_i2s.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 99a128a666fb..c9fedf6eb2e6 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -808,8 +808,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 
 	i2s->bclk_ratio = 64;
 	i2s->pinctrl = devm_pinctrl_get(&pdev->dev);
-	if (IS_ERR(i2s->pinctrl))
+	if (IS_ERR(i2s->pinctrl)) {
 		dev_err(&pdev->dev, "failed to find i2s pinctrl\n");
+		ret = PTR_ERR(i2s->pinctrl);
+		goto err_clk;
+	}
 
 	i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl,
 				   "bclk_on");
-- 
cgit v1.2.3