From b9a4b57f423ff8ff9ab9b68a238bdc3e7678f723 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Wed, 21 Jul 2021 17:04:45 +0200
Subject: ASoC: codecs: wcd938x: fix wcd module dependency

With SND_SOC_ALL_CODECS=y and SND_SOC_WCD938X_SDW=m, there is a link
error from a reverse dependency, since the built-in codec driver calls
into the modular soundwire back-end:

x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_free':
wcd938x.c:(.text+0x2c0): undefined reference to `wcd938x_sdw_free'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_hw_params':
wcd938x.c:(.text+0x2f6): undefined reference to `wcd938x_sdw_hw_params'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_set_sdw_stream':
wcd938x.c:(.text+0x332): undefined reference to `wcd938x_sdw_set_sdw_stream'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_tx_swr_ctrl':
wcd938x.c:(.text+0x23de): undefined reference to `wcd938x_swr_get_current_bank'
x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_bind':
wcd938x.c:(.text+0x2579): undefined reference to `wcd938x_sdw_device_get'
x86_64-linux-ld: wcd938x.c:(.text+0x25a1): undefined reference to `wcd938x_sdw_device_get'
x86_64-linux-ld: wcd938x.c:(.text+0x262a): undefined reference to `__devm_regmap_init_sdw'

Work around this using two small hacks: An added Kconfig dependency
prevents the main driver from being built-in when soundwire support
itself is a loadable module to allow calling devm_regmap_init_sdw(),
and a Makefile trick links the wcd938x-sdw backend as built-in
if needed to solve the dependency between the two modules.

Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210721150510.1837221-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig  | 1 +
 sound/soc/codecs/Makefile | 5 ++++-
 2 files changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 032c87637f63..d24edb01c19b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1559,6 +1559,7 @@ config SND_SOC_WCD934X
 config SND_SOC_WCD938X
 	depends on SND_SOC_WCD938X_SDW
 	tristate
+	depends on SOUNDWIRE || !SOUNDWIRE
 
 config SND_SOC_WCD938X_SDW
 	tristate "WCD9380/WCD9385 Codec - SDW"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index de8b83dd2c76..7bb38c370842 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -583,7 +583,10 @@ obj-$(CONFIG_SND_SOC_WCD_MBHC)	+= snd-soc-wcd-mbhc.o
 obj-$(CONFIG_SND_SOC_WCD9335)	+= snd-soc-wcd9335.o
 obj-$(CONFIG_SND_SOC_WCD934X)	+= snd-soc-wcd934x.o
 obj-$(CONFIG_SND_SOC_WCD938X)	+= snd-soc-wcd938x.o
-obj-$(CONFIG_SND_SOC_WCD938X_SDW) += snd-soc-wcd938x-sdw.o
+ifdef CONFIG_SND_SOC_WCD938X_SDW
+# avoid link failure by forcing sdw code built-in when needed
+obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x-sdw.o
+endif
 obj-$(CONFIG_SND_SOC_WL1273)	+= snd-soc-wl1273.o
 obj-$(CONFIG_SND_SOC_WM0010)	+= snd-soc-wm0010.o
 obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
-- 
cgit v1.2.3


From 6d20bf7c020f417fdef1810a22da17c126603472 Mon Sep 17 00:00:00 2001
From: Derek Fang <derek.fang@realtek.com>
Date: Wed, 21 Jul 2021 21:31:21 +0800
Subject: ASoC: rt5682: Adjust headset volume button threshold

Adjust the threshold of headset button volume+ to fix
the wrong button detection issue with some brand headsets.

Signed-off-by: Derek Fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/20210721133121.12333-1-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5682.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index abcd6f483788..51ecaa2abcd1 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = {
 	{RT5682_I2C_CTRL, 0x000f},
 	{RT5682_PLL2_INTERNAL, 0x8266},
 	{RT5682_SAR_IL_CMD_3, 0x8365},
+	{RT5682_SAR_IL_CMD_6, 0x0180},
 };
 
 void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev)
-- 
cgit v1.2.3


From d00f541a49406afc2c091aac121e29b3b61480a2 Mon Sep 17 00:00:00 2001
From: Mario Limonciello <mario.limonciello@amd.com>
Date: Wed, 21 Jul 2021 13:36:03 -0500
Subject: ASoC: amd: renoir: Run hibernation callbacks

The registers need to be re-initialized after hibernation or
microphone may be non-functional.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213793
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://lore.kernel.org/r/20210721183603.747-2-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/renoir/rn-pci-acp3x.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c
index 19438da5dfa5..7b8040e812a1 100644
--- a/sound/soc/amd/renoir/rn-pci-acp3x.c
+++ b/sound/soc/amd/renoir/rn-pci-acp3x.c
@@ -382,6 +382,8 @@ static const struct dev_pm_ops rn_acp_pm = {
 	.runtime_resume =  snd_rn_acp_resume,
 	.suspend = snd_rn_acp_suspend,
 	.resume =	snd_rn_acp_resume,
+	.restore =	snd_rn_acp_resume,
+	.poweroff =	snd_rn_acp_suspend,
 };
 
 static void snd_rn_acp_remove(struct pci_dev *pci)
-- 
cgit v1.2.3


From 090c57da5fd59fb59adc9d9341a77558c93b0abd Mon Sep 17 00:00:00 2001
From: Marek Vasut <marex@denx.de>
Date: Tue, 20 Jul 2021 22:03:48 +0200
Subject: ASoC: tlv320aic32x4: Fix TAS2505/TAS2521 processing block selection

The TAS2505/TAS2521 does support only three processing block options, unlike
TLV320AIC32x4 which supports 25. This is documented in TI slau472 2.5.1.2
Processing Blocks and Page 0 / Register 60: DAC Instruction Set - 0x00 / 0x3C.

Limit the Processing Blocks maximum value to 3 on TAS2505/TAS2521 and select
processing block PRB_P1 always, because for the configuration of teh codec
implemented in this driver, this is the best quality option.

Fixes: b4525b6196cd7 ("ASoC: tlv320aic32x4: add support for TAS2505")
Signed-off-by: Marek Vasut <marex@denx.de>
Cc: Claudius Heine <ch@denx.de>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210720200348.182139-1-marex@denx.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic32x4.c | 33 ++++++++++++++++++++++++++-------
 1 file changed, 26 insertions(+), 7 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index dcd8aeb45cb3..2e9175b37dc9 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -682,11 +682,20 @@ static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr)
 static int aic32x4_set_processing_blocks(struct snd_soc_component *component,
 						u8 r_block, u8 p_block)
 {
-	if (r_block > 18 || p_block > 25)
-		return -EINVAL;
+	struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
+
+	if (aic32x4->type == AIC32X4_TYPE_TAS2505) {
+		if (r_block || p_block > 3)
+			return -EINVAL;
 
-	snd_soc_component_write(component, AIC32X4_ADCSPB, r_block);
-	snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+		snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+	} else { /* AIC32x4 */
+		if (r_block > 18 || p_block > 25)
+			return -EINVAL;
+
+		snd_soc_component_write(component, AIC32X4_ADCSPB, r_block);
+		snd_soc_component_write(component, AIC32X4_DACSPB, p_block);
+	}
 
 	return 0;
 }
@@ -695,6 +704,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
 				unsigned int sample_rate, unsigned int channels,
 				unsigned int bit_depth)
 {
+	struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component);
 	u8 aosr;
 	u16 dosr;
 	u8 adc_resource_class, dac_resource_class;
@@ -721,19 +731,28 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component,
 		adc_resource_class = 6;
 		dac_resource_class = 8;
 		dosr_increment = 8;
-		aic32x4_set_processing_blocks(component, 1, 1);
+		if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+			aic32x4_set_processing_blocks(component, 0, 1);
+		else
+			aic32x4_set_processing_blocks(component, 1, 1);
 	} else if (sample_rate <= 96000) {
 		aosr = 64;
 		adc_resource_class = 6;
 		dac_resource_class = 8;
 		dosr_increment = 4;
-		aic32x4_set_processing_blocks(component, 1, 9);
+		if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+			aic32x4_set_processing_blocks(component, 0, 1);
+		else
+			aic32x4_set_processing_blocks(component, 1, 9);
 	} else if (sample_rate == 192000) {
 		aosr = 32;
 		adc_resource_class = 3;
 		dac_resource_class = 4;
 		dosr_increment = 2;
-		aic32x4_set_processing_blocks(component, 13, 19);
+		if (aic32x4->type == AIC32X4_TYPE_TAS2505)
+			aic32x4_set_processing_blocks(component, 0, 1);
+		else
+			aic32x4_set_processing_blocks(component, 13, 19);
 	} else {
 		dev_err(component->dev, "Sampling rate not supported\n");
 		return -EINVAL;
-- 
cgit v1.2.3


From 5434d0dc56bce4510109a431a7eb71ec5131ef0f Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Thu, 22 Jul 2021 18:33:15 +0530
Subject: ASoC: amd: enable stop_dma_first flag for cz_dai_7219_98357 dai link

DMA driver stop sequence should be invoked first before invoking I2S
controller driver stop sequence for Stoneyridge platform.

Enable stop_dma_first flag for cz_dai_7219_98357 dai link structure.

Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20210722130328.23796-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/acp-da7219-max98357a.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 9449fb40a956..3c60c5f96dcb 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -525,6 +525,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_da7219_play_ops,
 		SND_SOC_DAILINK_REG(designware1, dlgs, platform),
 	},
@@ -534,6 +535,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_da7219_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, dlgs, platform),
 	},
@@ -543,6 +545,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_playback = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_max_play_ops,
 		SND_SOC_DAILINK_REG(designware3, mx, platform),
 	},
@@ -553,6 +556,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_dmic0_cap_ops,
 		SND_SOC_DAILINK_REG(designware3, adau, platform),
 	},
@@ -563,6 +567,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_dmic1_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, adau, platform),
 	},
-- 
cgit v1.2.3


From 2c39ca6885a2ec03e5c9e7c12a4da2aa8926605a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Fri, 23 Jul 2021 19:02:00 +0100
Subject: ASoC: tlv320aic31xx: Fix jack detection after suspend

The tlv320aic31xx driver relies on regcache_sync() to restore the register
contents after going to _BIAS_OFF, for example during system suspend. This
does not work for the jack detection configuration since that is configured
via the same register that status is read back from so the register is
volatile and not cached. This can also cause issues during init if the jack
detection ends up getting set up before the CODEC is initially brought out
of _BIAS_OFF, we will reset the CODEC and resync the cache as part of that
process.

Fix this by explicitly reapplying the jack detection configuration after
resyncing the register cache during power on.

This issue was found by an engineer working off-list on a product
kernel, I just wrote up the upstream fix.

Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210723180200.25105-1-broonie@kernel.org
Cc: stable@vger.kernel.org
---
 sound/soc/codecs/tlv320aic31xx.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index b504d63385b3..52d2c968b5c0 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -35,6 +35,9 @@
 
 #include "tlv320aic31xx.h"
 
+static int aic31xx_set_jack(struct snd_soc_component *component,
+                            struct snd_soc_jack *jack, void *data);
+
 static const struct reg_default aic31xx_reg_defaults[] = {
 	{ AIC31XX_CLKMUX, 0x00 },
 	{ AIC31XX_PLLPR, 0x11 },
@@ -1256,6 +1259,13 @@ static int aic31xx_power_on(struct snd_soc_component *component)
 		return ret;
 	}
 
+	/*
+	 * The jack detection configuration is in the same register
+	 * that is used to report jack detect status so is volatile
+	 * and not covered by the cache sync, restore it separately.
+	 */
+	aic31xx_set_jack(component, aic31xx->jack, NULL);
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From 0f32d9eb38c13c32895b5bf695eac639cee02d6c Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Mon, 26 Jul 2021 17:45:25 +0800
Subject: ASoC: Intel: sof_da7219_mx98360a: fail to initialize soundcard

The default codec for speaker amp's DAI Link is max98373 and will be
overwritten in probe function if the board id is sof_da7219_mx98360a.
However, the probe function does not do it because the board id is
changed in earlier commit.

Fixes: 1cc04d195dc2 ("ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters")
Signed-off-by: Brent Lu <brent.lu@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210726094525.5748-1-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/sof_da7219_max98373.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c
index 896251d742fe..b7b3b0bf994a 100644
--- a/sound/soc/intel/boards/sof_da7219_max98373.c
+++ b/sound/soc/intel/boards/sof_da7219_max98373.c
@@ -404,7 +404,7 @@ static int audio_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	/* By default dais[0] is configured for max98373 */
-	if (!strcmp(pdev->name, "sof_da7219_max98360a")) {
+	if (!strcmp(pdev->name, "sof_da7219_mx98360a")) {
 		dais[0] = (struct snd_soc_dai_link) {
 			.name = "SSP1-Codec",
 			.id = 0,
-- 
cgit v1.2.3


From 2635c226036c1bf44b86575d0bc721505c8201e3 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Mon, 26 Jul 2021 13:21:42 -0500
Subject: ASoC: topology: Select SND_DYNAMIC_MINORS

The indexes of the devices are described within the topology file, it is a
possibility that the topology encodes invalid indexes when DYNAMIC_MINORS
is not enabled in kernel:

 #define SNDRV_MINOR_COMPRESS		2	/* 2 - 3 */
 #define SNDRV_MINOR_HWDEP		4	/* 4 - 7 */
 #define SNDRV_MINOR_RAWMIDI		8	/* 8 - 15 */
 #define SNDRV_MINOR_PCM_PLAYBACK	16	/* 16 - 23 */
 #define SNDRV_MINOR_PCM_CAPTURE	24	/* 24 - 31 */

If the topology assigns an index greater than 7 for PLAYBACK/CAPTURE PCM
then there will be minor number collision.

As an example:
card0 creates a capture PCM with index 10 -> minor = 34
card1 creates compress device with index 0 -> minor = 34

Card1 will fail to instantiate because the minor for the compress stream is
already taken.

To avoid seemingly mysterious issues with card creation, select the
DYNAMIC_MINORS when the topology is enabled.

The other option would be to try to do out of bound index checks in case of
DYNAMIC_MINOR is not enabled and do not even attempt to create the device
with failing the topology load.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210726182142.179604-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/Kconfig | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/soc')

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 8a13462e1a63..5dcf77af07af 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_COMPRESS
 
 config SND_SOC_TOPOLOGY
 	bool
+	select SND_DYNAMIC_MINORS
 
 config SND_SOC_TOPOLOGY_KUNIT_TEST
 	tristate "KUnit tests for SoC topology"
-- 
cgit v1.2.3


From 61bef9e68dca4316e1fc5fdf5f0c270bdbd65657 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 26 Jul 2021 13:28:55 -0500
Subject: ASoC: SOF: Intel: hda: enforce exclusion between HDaudio and
 SoundWire

On some platforms with an external HDaudio codec, the DSDT reports the
presence of SoundWire devices. Pin-mux restrictions and board reworks
usually prevent coexistence between the two types of links, let's
prevent unnecessary operations from starting.

In the case of a single iDISP codec being detected, we still start the
links even if no SoundWire machine configuration was detected, so that
we can double-check what the hardware is and add the missing
configuration if applicable.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Link: https://lore.kernel.org/r/20210726182855.179943-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index e1e368ff2b12..891e6e1b9121 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -187,12 +187,16 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev)
 int hda_sdw_startup(struct snd_sof_dev *sdev)
 {
 	struct sof_intel_hda_dev *hdev;
+	struct snd_sof_pdata *pdata = sdev->pdata;
 
 	hdev = sdev->pdata->hw_pdata;
 
 	if (!hdev->sdw)
 		return 0;
 
+	if (pdata->machine && !pdata->machine->mach_params.link_mask)
+		return 0;
+
 	return sdw_intel_startup(hdev->sdw);
 }
 
@@ -1002,6 +1006,14 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev)
 			hda_mach->mach_params.dmic_num = dmic_num;
 			pdata->machine = hda_mach;
 			pdata->tplg_filename = tplg_filename;
+
+			if (codec_num == 2) {
+				/*
+				 * Prevent SoundWire links from starting when an external
+				 * HDaudio codec is used
+				 */
+				hda_mach->mach_params.link_mask = 0;
+			}
 		}
 	}
 
-- 
cgit v1.2.3


From 31428c78748cafdd9352e1f622eb89bf453d9700 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Mon, 26 Jul 2021 20:41:23 +0100
Subject: ASoC: component: Remove misplaced prefix handling in pin control
 functions

When the component level pin control functions were added they for some
no longer obvious reason handled adding prefixing of widget names. This
meant that when the lack of prefix handling in the DAPM level pin
operations was fixed by ae4fc532244b3bb4d (ASoC: dapm: use component
prefix when checking widget names) the one device using the component
level API ended up with the prefix being applied twice, causing all
lookups to fail.

Fix this by removing the redundant prefixing from the component code,
which has the nice side effect of also making that code much simpler.

Reported-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tested-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210726194123.54585-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-component.c | 63 ++++++++++++++++++++---------------------------
 1 file changed, 27 insertions(+), 36 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 3a5e84e16a87..c8dfd0de30e4 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -148,86 +148,75 @@ int snd_soc_component_set_bias_level(struct snd_soc_component *component,
 	return soc_component_ret(component, ret);
 }
 
-static int soc_component_pin(struct snd_soc_component *component,
-			     const char *pin,
-			     int (*pin_func)(struct snd_soc_dapm_context *dapm,
-					     const char *pin))
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix) {
-		ret = pin_func(dapm, pin);
-		goto end;
-	}
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name) {
-		ret = -ENOMEM;
-		goto end;
-	}
-
-	ret = pin_func(dapm, full_name);
-	kfree(full_name);
-end:
-	return soc_component_ret(component, ret);
-}
-
 int snd_soc_component_enable_pin(struct snd_soc_component *component,
 				 const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_enable_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_enable_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin);
 
 int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component,
 					  const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_enable_pin_unlocked);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_enable_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin_unlocked);
 
 int snd_soc_component_disable_pin(struct snd_soc_component *component,
 				  const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_disable_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_disable_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin);
 
 int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component,
 					   const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_disable_pin_unlocked);
+	struct snd_soc_dapm_context *dapm = 
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_disable_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin_unlocked);
 
 int snd_soc_component_nc_pin(struct snd_soc_component *component,
 			     const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_nc_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_nc_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin);
 
 int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component,
 				      const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_nc_pin_unlocked);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_nc_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin_unlocked);
 
 int snd_soc_component_get_pin_status(struct snd_soc_component *component,
 				     const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_get_pin_status);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_get_pin_status(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_get_pin_status);
 
 int snd_soc_component_force_enable_pin(struct snd_soc_component *component,
 				       const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_force_enable_pin(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin);
 
@@ -235,7 +224,9 @@ int snd_soc_component_force_enable_pin_unlocked(
 	struct snd_soc_component *component,
 	const char *pin)
 {
-	return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin_unlocked);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	return snd_soc_dapm_force_enable_pin_unlocked(dapm, pin);
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked);
 
-- 
cgit v1.2.3


From acbf58e530416e167c3b323111f4013d9f2b0a7d Mon Sep 17 00:00:00 2001
From: Lucas Tanure <tanureal@opensource.cirrus.com>
Date: Wed, 28 Jul 2021 11:44:16 +0100
Subject: ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs

soc_cleanup_component_debugfs will debugfs_remove_recursive
the component->debugfs_root, so adsp doesn't need to also
remove the same entry.
By doing that adsp also creates a race with core component,
which causes a NULL pointer dereference

Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210728104416.636591-1-tanureal@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm_adsp.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 549d98241dae..fe15cbc7bcaf 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -747,7 +747,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
 static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp)
 {
 	wm_adsp_debugfs_clear(dsp);
-	debugfs_remove_recursive(dsp->debugfs_root);
 }
 #else
 static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp,
-- 
cgit v1.2.3


From ee86f680ff4c9b406d49d4e22ddf10805b8a2137 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 29 Jul 2021 18:09:27 +0100
Subject: ASoC: cs42l42: Correct definition of ADC Volume control

The ADC volume is a signed 8-bit number with range -97 to +12,
with -97 being mute. Use a SOC_SINGLE_S8_TLV() to define this
and fix the DECLARE_TLV_DB_SCALE() to have the correct start and
mute flag.

Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210729170929.6589-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index eff013f295be..2fd20511f246 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = {
 	.use_single_write = true,
 };
 
-static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
+static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true);
 static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
 
 static const char * const cs42l42_hpf_freq_text[] = {
@@ -444,8 +444,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 				CS42L42_ADC_INV_SHIFT, true, false),
 	SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL,
 				CS42L42_ADC_DIG_BOOST_SHIFT, true, false),
-	SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME,
-				CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv),
+	SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv),
 	SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL,
 				CS42L42_ADC_WNF_EN_SHIFT, true, false),
 	SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL,
-- 
cgit v1.2.3


From 64324bac750b84ca54711fb7d332132fcdb87293 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 29 Jul 2021 18:09:28 +0100
Subject: ASoC: cs42l42: Don't allow SND_SOC_DAIFMT_LEFT_J

The driver has no support for left-justified protocol so it should
not have been allowing this to be passed to cs42l42_set_dai_fmt().

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210729170929.6589-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 2fd20511f246..38e243a815b1 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -788,7 +788,6 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-	case SND_SOC_DAIFMT_LEFT_J:
 		break;
 	default:
 		return -EINVAL;
-- 
cgit v1.2.3


From 926ef1a4c245c093acc07807e466ad2ef0ff6ccb Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 29 Jul 2021 18:09:29 +0100
Subject: ASoC: cs42l42: Fix bclk calculation for mono

An I2S frame always has a left and right channel slot even if mono
data is being sent. So if channels==1 the actual bitclock frequency
is 2 * snd_soc_params_to_bclk(params).

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2cdba9b045c7 ("ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called")
Link: https://lore.kernel.org/r/20210729170929.6589-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 38e243a815b1..08ca05bfbeb3 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -830,6 +830,10 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 	cs42l42->srate = params_rate(params);
 	cs42l42->bclk = snd_soc_params_to_bclk(params);
 
+	/* I2S frame always has 2 channels even for mono audio */
+	if (channels == 1)
+		cs42l42->bclk *= 2;
+
 	switch(substream->stream) {
 	case SNDRV_PCM_STREAM_CAPTURE:
 		if (channels == 2) {
-- 
cgit v1.2.3


From 2e6b836312a477d647a7920b56810a5a25f6c856 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 28 Jul 2021 13:23:50 +0200
Subject: ASoC: intel: atom: Fix reference to PCM buffer address

PCM buffers might be allocated dynamically when the buffer
preallocation failed or a larger buffer is requested, and it's not
guaranteed that substream->dma_buffer points to the actually used
buffer.  The address should be retrieved from runtime->dma_addr,
instead of substream->dma_buffer (and shouldn't use virt_to_phys).

Also, remove the line overriding runtime->dma_area superfluously,
which was already set up at the PCM buffer allocation.

Cc: Cezary Rojewski <cezary.rojewski@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/atom/sst-mfld-platform-pcm.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 4124aa2fc247..5db2f4865bbb 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream,
 	snd_pcm_uframes_t period_size;
 	ssize_t periodbytes;
 	ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
-	u32 buffer_addr = virt_to_phys(substream->dma_buffer.area);
+	u32 buffer_addr = substream->runtime->dma_addr;
 
 	channels = substream->runtime->channels;
 	period_size = substream->runtime->period_size;
@@ -233,7 +233,6 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
 	/* set codec params and inform SST driver the same */
 	sst_fill_pcm_params(substream, &param);
 	sst_fill_alloc_params(substream, &alloc_params);
-	substream->runtime->dma_area = substream->dma_buffer.area;
 	str_params.sparams = param;
 	str_params.aparams = alloc_params;
 	str_params.codec = SST_CODEC_TYPE_PCM;
-- 
cgit v1.2.3


From 42bc62c9f1d3d4880bdc27acb5ab4784209bb0b0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 28 Jul 2021 13:23:51 +0200
Subject: ASoC: xilinx: Fix reference to PCM buffer address

PCM buffers might be allocated dynamically when the buffer
preallocation failed or a larger buffer is requested, and it's not
guaranteed that substream->dma_buffer points to the actually used
buffer.  The driver needs to refer to substream->runtime->dma_addr
instead for the buffer address.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/xilinx/xlnx_formatter_pcm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index 1d59fb668c77..91afea9d5de6 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -452,8 +452,8 @@ static int xlnx_formatter_pcm_hw_params(struct snd_soc_component *component,
 
 	stream_data->buffer_size = size;
 
-	low = lower_32_bits(substream->dma_buffer.addr);
-	high = upper_32_bits(substream->dma_buffer.addr);
+	low = lower_32_bits(runtime->dma_addr);
+	high = upper_32_bits(runtime->dma_addr);
 	writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB);
 	writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB);
 
-- 
cgit v1.2.3


From 827f3164aaa579eee6fd50c6654861d54f282a11 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 28 Jul 2021 13:23:52 +0200
Subject: ASoC: uniphier: Fix reference to PCM buffer address

Along with the transition to the managed PCM buffers, the driver now
accepts the dynamically allocated buffer, while it still kept the
reference to the old preallocated buffer address.  This patch corrects
to the right reference via runtime->dma_addr.

(Although this might have been already buggy before the cleanup with
the managed buffer, let's put Fixes tag to point that; it's a corner
case, after all.)

Fixes: d55894bc2763 ("ASoC: uniphier: Use managed buffer allocation")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-5-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/uniphier/aio-dma.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c
index 3c1628a3a1ac..3d9736e7381f 100644
--- a/sound/soc/uniphier/aio-dma.c
+++ b/sound/soc/uniphier/aio-dma.c
@@ -198,7 +198,7 @@ static int uniphier_aiodma_mmap(struct snd_soc_component *component,
 	vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot);
 
 	return remap_pfn_range(vma, vma->vm_start,
-			       substream->dma_buffer.addr >> PAGE_SHIFT,
+			       substream->runtime->dma_addr >> PAGE_SHIFT,
 			       vma->vm_end - vma->vm_start, vma->vm_page_prot);
 }
 
-- 
cgit v1.2.3


From bb6a40fc5a830cae45ddd5cd6cfa151b008522ed Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 28 Jul 2021 13:23:53 +0200
Subject: ASoC: kirkwood: Fix reference to PCM buffer address

The transition to the managed PCM buffers allowed the dynamically
buffer allocation, while the driver code still assumes the fixed
preallocation buffer and sets up the DMA stuff at the open call.
This needs to be moved to hw_params after the buffer allocation and
setup.  Also, the reference to the buffer address has to be corrected
to runtime->dma_addr.

Fixes: b3c0ae75f5d3 ("ASoC: kirkwood: Use managed DMA buffer allocation")
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210728112353.6675-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/kirkwood/kirkwood-dma.c | 26 ++++++++++++++++++--------
 1 file changed, 18 insertions(+), 8 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index c2a5933bfcfc..700a18561a94 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -104,8 +104,6 @@ static int kirkwood_dma_open(struct snd_soc_component *component,
 	int err;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
-	const struct mbus_dram_target_info *dram;
-	unsigned long addr;
 
 	snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
 
@@ -142,20 +140,14 @@ static int kirkwood_dma_open(struct snd_soc_component *component,
 		writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK);
 	}
 
-	dram = mv_mbus_dram_info();
-	addr = substream->dma_buffer.addr;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (priv->substream_play)
 			return -EBUSY;
 		priv->substream_play = substream;
-		kirkwood_dma_conf_mbus_windows(priv->io,
-			KIRKWOOD_PLAYBACK_WIN, addr, dram);
 	} else {
 		if (priv->substream_rec)
 			return -EBUSY;
 		priv->substream_rec = substream;
-		kirkwood_dma_conf_mbus_windows(priv->io,
-			KIRKWOOD_RECORD_WIN, addr, dram);
 	}
 
 	return 0;
@@ -182,6 +174,23 @@ static int kirkwood_dma_close(struct snd_soc_component *component,
 	return 0;
 }
 
+static int kirkwood_dma_hw_params(struct snd_soc_component *component,
+				  struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params)
+{
+	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
+	const struct mbus_dram_target_info *dram = mv_mbus_dram_info();
+	unsigned long addr = substream->runtime->dma_addr;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		kirkwood_dma_conf_mbus_windows(priv->io,
+			KIRKWOOD_PLAYBACK_WIN, addr, dram);
+	else
+		kirkwood_dma_conf_mbus_windows(priv->io,
+			KIRKWOOD_RECORD_WIN, addr, dram);
+	return 0;
+}
+
 static int kirkwood_dma_prepare(struct snd_soc_component *component,
 				struct snd_pcm_substream *substream)
 {
@@ -246,6 +255,7 @@ const struct snd_soc_component_driver kirkwood_soc_component = {
 	.name		= DRV_NAME,
 	.open		= kirkwood_dma_open,
 	.close		= kirkwood_dma_close,
+	.hw_params	= kirkwood_dma_hw_params,
 	.prepare	= kirkwood_dma_prepare,
 	.pointer	= kirkwood_dma_pointer,
 	.pcm_construct	= kirkwood_dma_new,
-- 
cgit v1.2.3


From 1d25684e22516f1cff77176d288b1da02fff57bb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Thu, 29 Jul 2021 00:47:29 +0100
Subject: ASoC: nau8824: Fix open coded prefix handling

As with the component layer code the nau8824 driver had been doing some
open coded pin manipulation which will have been broken now the core is
fixed to handle this properly, remove the open coding to avoid the issue.

Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210728234729.10135-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/nau8824.c | 42 ++++++------------------------------------
 1 file changed, 6 insertions(+), 36 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 15bd8335f667..db88be48c998 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap)
 	}
 }
 
-static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin)
-{
-	struct snd_soc_dapm_context *dapm = nau8824->dapm;
-	const char *prefix = dapm->component->name_prefix;
-	char prefixed_pin[80];
-
-	if (prefix) {
-		snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
-			 prefix, pin);
-		snd_soc_dapm_disable_pin(dapm, prefixed_pin);
-	} else {
-		snd_soc_dapm_disable_pin(dapm, pin);
-	}
-}
-
-static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin)
-{
-	struct snd_soc_dapm_context *dapm = nau8824->dapm;
-	const char *prefix = dapm->component->name_prefix;
-	char prefixed_pin[80];
-
-	if (prefix) {
-		snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s",
-			 prefix, pin);
-		snd_soc_dapm_force_enable_pin(dapm, prefixed_pin);
-	} else {
-		snd_soc_dapm_force_enable_pin(dapm, pin);
-	}
-}
-
 static void nau8824_eject_jack(struct nau8824 *nau8824)
 {
 	struct snd_soc_dapm_context *dapm = nau8824->dapm;
@@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824)
 	/* Clear all interruption status */
 	nau8824_int_status_clear_all(regmap);
 
-	nau8824_dapm_disable_pin(nau8824, "SAR");
-	nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+	snd_soc_dapm_disable_pin(dapm, "SAR");
+	snd_soc_dapm_disable_pin(dapm, "MICBIAS");
 	snd_soc_dapm_sync(dapm);
 
 	/* Enable the insertion interruption, disable the ejection
@@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work)
 	struct regmap *regmap = nau8824->regmap;
 	int adc_value, event = 0, event_mask = 0;
 
-	nau8824_dapm_enable_pin(nau8824, "MICBIAS");
-	nau8824_dapm_enable_pin(nau8824, "SAR");
+	snd_soc_dapm_enable_pin(dapm, "MICBIAS");
+	snd_soc_dapm_enable_pin(dapm, "SAR");
 	snd_soc_dapm_sync(dapm);
 
 	msleep(100);
@@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work)
 	if (adc_value < HEADSET_SARADC_THD) {
 		event |= SND_JACK_HEADPHONE;
 
-		nau8824_dapm_disable_pin(nau8824, "SAR");
-		nau8824_dapm_disable_pin(nau8824, "MICBIAS");
+		snd_soc_dapm_disable_pin(dapm, "SAR");
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS");
 		snd_soc_dapm_sync(dapm);
 	} else {
 		event |= SND_JACK_HEADSET;
-- 
cgit v1.2.3


From 8b5d95313b6d30f642e4ed0125891984c446604e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 31 Jul 2021 10:43:31 +0200
Subject: ASoC: amd: Fix reference to PCM buffer address

PCM buffers might be allocated dynamically when the buffer
preallocation failed or a larger buffer is requested, and it's not
guaranteed that substream->dma_buffer points to the actually used
buffer.  The driver needs to refer to substream->runtime->dma_addr
instead for the buffer address.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210731084331.32225-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/acp-pcm-dma.c          | 2 +-
 sound/soc/amd/raven/acp3x-pcm-dma.c  | 2 +-
 sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 143155a840ac..cc1ce6f22caa 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -969,7 +969,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component,
 
 	acp_set_sram_bank_state(rtd->acp_mmio, 0, true);
 	/* Save for runtime private data */
-	rtd->dma_addr = substream->dma_buffer.addr;
+	rtd->dma_addr = runtime->dma_addr;
 	rtd->order = get_order(size);
 
 	/* Fill the page table entries in ACP SRAM */
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index 8148b0d22e88..597d7c4b2a6b 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -286,7 +286,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component,
 		pr_err("pinfo failed\n");
 	}
 	size = params_buffer_bytes(params);
-	rtd->dma_addr = substream->dma_buffer.addr;
+	rtd->dma_addr = substream->runtime->dma_addr;
 	rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 	config_acp3x_dma(rtd, substream->stream);
 	return 0;
diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c
index bd20622b0933..0391c28dd078 100644
--- a/sound/soc/amd/renoir/acp3x-pdm-dma.c
+++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c
@@ -242,7 +242,7 @@ static int acp_pdm_dma_hw_params(struct snd_soc_component *component,
 		return -EINVAL;
 	size = params_buffer_bytes(params);
 	period_bytes = params_period_bytes(params);
-	rtd->dma_addr = substream->dma_buffer.addr;
+	rtd->dma_addr = substream->runtime->dma_addr;
 	rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 	config_acp_dma(rtd, substream->stream);
 	init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes,
-- 
cgit v1.2.3


From 6b994c554ebc4c065427f510db333081cbd7228d Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 2 Aug 2021 10:16:28 -0500
Subject: ASoC: SOF: Intel: Kconfig: fix SoundWire dependencies

The previous Kconfig cleanup added simplifications but also introduced
a new one by moving a boolean to a tristate. This leads to randconfig
problems.

This patch moves the select operations in the SOUNDWIRE_LINK_BASELINE
option. The INTEL_SOUNDWIRE config remains a tristate for backwards
compatibility with older configurations but is essentially an on/off
switch.

Fixes: cf5807f5f814f ('ASoC: SOF: Intel: SoundWire: simplify Kconfig')
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Tested-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210802151628.15291-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/Kconfig | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 4bce89b5ea40..4447f515e8b1 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -278,6 +278,8 @@ config SND_SOC_SOF_HDA
 
 config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
 	tristate
+	select SOUNDWIRE_INTEL if SND_SOC_SOF_INTEL_SOUNDWIRE
+	select SND_INTEL_SOUNDWIRE_ACPI if SND_SOC_SOF_INTEL_SOUNDWIRE
 
 config SND_SOC_SOF_INTEL_SOUNDWIRE
 	tristate "SOF support for SoundWire"
@@ -285,8 +287,6 @@ config SND_SOC_SOF_INTEL_SOUNDWIRE
 	depends on SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE
 	depends on ACPI && SOUNDWIRE
 	depends on !(SOUNDWIRE=m && SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE=y)
-	select SOUNDWIRE_INTEL
-	select SND_INTEL_SOUNDWIRE_ACPI
 	help
 	  This adds support for SoundWire with Sound Open Firmware
 	  for Intel(R) platforms.
-- 
cgit v1.2.3


From 973b393fdf073a4ebd8d82ef6edea99fedc74af9 Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Date: Mon, 2 Aug 2021 10:17:49 -0500
Subject: ASoC: SOF: Intel: hda-ipc: fix reply size checking

Checking that two values don't have common bits makes no sense,
strict equality is meant.

Fixes: f3b433e4699f  ("ASoC: SOF: Implement Probe IPC API")
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210802151749.15417-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-ipc.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index c91aa951df22..acfeca42604c 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -107,8 +107,8 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev)
 	} else {
 		/* reply correct size ? */
 		if (reply.hdr.size != msg->reply_size &&
-			/* getter payload is never known upfront */
-			!(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) {
+		    /* getter payload is never known upfront */
+		    ((reply.hdr.cmd & SOF_GLB_TYPE_MASK) != SOF_IPC_GLB_PROBE)) {
 			dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n",
 				msg->reply_size, reply.hdr.size);
 			ret = -EINVAL;
-- 
cgit v1.2.3


From 30615bd21b4cc3c3bb5ae8bd70e2a915cc5f75c7 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Tue, 3 Aug 2021 17:08:33 +0100
Subject: ASoC: cs42l42: Fix inversion of ADC Notch Switch control

The underlying register field has inverted sense (0 = enabled) so
the control definition must be marked as inverted.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210803160834.9005-1-rf@opensource.cirrus.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/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 08ca05bfbeb3..2db87e3a4dc3 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -437,7 +437,7 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
 static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 	/* ADC Volume and Filter Controls */
 	SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL,
-				CS42L42_ADC_NOTCH_DIS_SHIFT, true, false),
+				CS42L42_ADC_NOTCH_DIS_SHIFT, true, true),
 	SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL,
 				CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false),
 	SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL,
-- 
cgit v1.2.3


From 8b353bbeae20e2214c9d9d88bcb2fda4ba145d83 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Tue, 3 Aug 2021 17:08:34 +0100
Subject: ASoC: cs42l42: Remove duplicate control for WNF filter frequency

The driver was defining two ALSA controls that both change the same
register field for the wind noise filter corner frequency. The filter
response has two corners, at different frequencies, and the duplicate
controls most likely were an attempt to be able to set the value using
either of the frequencies.

However, having two controls changing the same field can be problematic
and it is unnecessary. Both frequencies are related to each other so
setting one implies exactly what the other would be.

Removing a control affects user-side code, but there is currently no
known use of the removed control so it would be best to remove it now
before it becomes a problem.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210803160834.9005-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 10 ----------
 1 file changed, 10 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 2db87e3a4dc3..a00dc3c65549 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -425,15 +425,6 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
 			    CS42L42_ADC_WNF_CF_SHIFT,
 			    cs42l42_wnf3_freq_text);
 
-static const char * const cs42l42_wnf05_freq_text[] = {
-	"280Hz", "315Hz", "350Hz", "385Hz",
-	"420Hz", "455Hz", "490Hz", "525Hz"
-};
-
-static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
-			    CS42L42_ADC_WNF_CF_SHIFT,
-			    cs42l42_wnf05_freq_text);
-
 static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 	/* ADC Volume and Filter Controls */
 	SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL,
@@ -451,7 +442,6 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
 				CS42L42_ADC_HPF_EN_SHIFT, true, false),
 	SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum),
 	SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum),
-	SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum),
 
 	/* DAC Volume and Filter Controls */
 	SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1,
-- 
cgit v1.2.3


From f1040e86f83b0f7d5f45724500a6a441731ff4b7 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 5 Aug 2021 17:11:04 +0100
Subject: ASoC: cs42l42: PLL must be running when changing MCLK_SRC_SEL

Both SCLK and PLL clocks must be running to drive the glitch-free mux
behind MCLK_SRC_SEL and complete the switchover.

This patch moves the writing of MCLK_SRC_SEL to when the PLL is started
and stopped, so that it only transitions while the PLL is running.
The unconditional write MCLK_SRC_SEL=0 in cs42l42_mute_stream() is safe
because if the PLL is not running MCLK_SRC_SEL is already 0.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 43fc357199f9 ("ASoC: cs42l42: Set clock source for both ways of stream")
Link: https://lore.kernel.org/r/20210805161111.10410-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 25 ++++++++++++++++++-------
 sound/soc/codecs/cs42l42.h |  1 +
 2 files changed, 19 insertions(+), 7 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index a00dc3c65549..c96549fe6ab2 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -619,6 +619,8 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 
 	for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
 		if (pll_ratio_table[i].sclk == clk) {
+			cs42l42->pll_config = i;
+
 			/* Configure the internal sample rate */
 			snd_soc_component_update_bits(component, CS42L42_MCLK_CTL,
 					CS42L42_INTERNAL_FS_MASK,
@@ -627,14 +629,9 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					(pll_ratio_table[i].mclk_int !=
 					24000000)) <<
 					CS42L42_INTERNAL_FS_SHIFT);
-			/* Set the MCLK src (PLL or SCLK) and the divide
-			 * ratio
-			 */
+
 			snd_soc_component_update_bits(component, CS42L42_MCLK_SRC_SEL,
-					CS42L42_MCLK_SRC_SEL_MASK |
 					CS42L42_MCLKDIV_MASK,
-					(pll_ratio_table[i].mclk_src_sel
-					<< CS42L42_MCLK_SRC_SEL_SHIFT) |
 					(pll_ratio_table[i].mclk_div <<
 					CS42L42_MCLKDIV_SHIFT));
 			/* Set up the LRCLK */
@@ -892,13 +889,21 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 			 */
 			regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_osc_seq,
 					       ARRAY_SIZE(cs42l42_to_osc_seq));
+
+			/* Must disconnect PLL before stopping it */
+			snd_soc_component_update_bits(component,
+						      CS42L42_MCLK_SRC_SEL,
+						      CS42L42_MCLK_SRC_SEL_MASK,
+						      0);
+			usleep_range(100, 200);
+
 			snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
 						      CS42L42_PLL_START_MASK, 0);
 		}
 	} else {
 		if (!cs42l42->stream_use) {
 			/* SCLK must be running before codec unmute */
-			if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) {
+			if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) {
 				snd_soc_component_update_bits(component, CS42L42_PLL_CTL1,
 							      CS42L42_PLL_START_MASK, 1);
 
@@ -919,6 +924,12 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 							       CS42L42_PLL_LOCK_TIMEOUT_US);
 				if (ret < 0)
 					dev_warn(component->dev, "PLL failed to lock: %d\n", ret);
+
+				/* PLL must be running to drive glitchless switch logic */
+				snd_soc_component_update_bits(component,
+							      CS42L42_MCLK_SRC_SEL,
+							      CS42L42_MCLK_SRC_SEL_MASK,
+							      CS42L42_MCLK_SRC_SEL_MASK);
 			}
 
 			/* Mark SCLK as present, turn off internal oscillator */
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 7bf05ff05f74..3dfeaf6611d1 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -775,6 +775,7 @@ struct  cs42l42_private {
 	struct gpio_desc *reset_gpio;
 	struct completion pdn_done;
 	struct snd_soc_jack *jack;
+	int pll_config;
 	int bclk;
 	u32 sclk;
 	u32 srate;
-- 
cgit v1.2.3


From 0c2f2ad4f16a58879463d0979a54293f8f296d6f Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 5 Aug 2021 17:11:05 +0100
Subject: ASoC: cs42l42: Fix LRCLK frame start edge

An I2S frame starts on the falling edge of LRCLK so ASP_STP must
be 0.

At the same time, move other format settings in the same register
from cs42l42_pll_config() to cs42l42_set_dai_fmt() where you'd
expect to find them, and merge into a single write.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20210805161111.10410-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 21 ++++++++++++---------
 1 file changed, 12 insertions(+), 9 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index c96549fe6ab2..02486329a570 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -667,15 +667,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component)
 					CS42L42_FSYNC_PULSE_WIDTH_MASK,
 					CS42L42_FRAC1_VAL(fsync - 1) <<
 					CS42L42_FSYNC_PULSE_WIDTH_SHIFT);
-			snd_soc_component_update_bits(component,
-					CS42L42_ASP_FRM_CFG,
-					CS42L42_ASP_5050_MASK,
-					CS42L42_ASP_5050_MASK);
-			/* Set the frame delay to 1.0 SCLK clocks */
-			snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG,
-					CS42L42_ASP_FSD_MASK,
-					CS42L42_ASP_FSD_1_0 <<
-					CS42L42_ASP_FSD_SHIFT);
 			/* Set the sample rates (96k or lower) */
 			snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN,
 					CS42L42_FS_EN_MASK,
@@ -775,6 +766,18 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
+		/*
+		 * 5050 mode, frame starts on falling edge of LRCLK,
+		 * frame delayed by 1.0 SCLKs
+		 */
+		snd_soc_component_update_bits(component,
+					      CS42L42_ASP_FRM_CFG,
+					      CS42L42_ASP_STP_MASK |
+					      CS42L42_ASP_5050_MASK |
+					      CS42L42_ASP_FSD_MASK,
+					      CS42L42_ASP_5050_MASK |
+					      (CS42L42_ASP_FSD_1_0 <<
+						CS42L42_ASP_FSD_SHIFT));
 		break;
 	default:
 		return -EINVAL;
-- 
cgit v1.2.3


From 3a5d89a9c6fe306d35dce4496abbb464c1454da0 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 5 Aug 2021 17:11:06 +0100
Subject: ASoC: cs42l42: Constrain sample rate to prevent illegal SCLK

The lowest valid SCLK corresponds to 44.1 kHz at 16-bit. Sample
rates less than this would produce SCLK below the minimum when using
a normal I2S frame. A constraint must be applied to prevent this.

The constraint is not applied if the machine driver sets SCLK, to
allow setups where the host generates additional bits per LRCLK
phase to increase the SCLK frequency. In these cases the machine
driver would always have to inform this driver of the actual SCLK,
and it must select a legal SCLK.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210805161111.10410-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 21 ++++++++++++++++++++-
 1 file changed, 20 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 02486329a570..29e0c8dc8466 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -807,6 +807,25 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	return 0;
 }
 
+static int cs42l42_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
+
+	/*
+	 * Sample rates < 44.1 kHz would produce an out-of-range SCLK with
+	 * a standard I2S frame. If the machine driver sets SCLK it must be
+	 * legal.
+	 */
+	if (cs42l42->sclk)
+		return 0;
+
+	/* Machine driver has not set a SCLK, limit bottom end to 44.1 kHz */
+	return snd_pcm_hw_constraint_minmax(substream->runtime,
+					    SNDRV_PCM_HW_PARAM_RATE,
+					    44100, 192000);
+}
+
 static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
@@ -966,8 +985,8 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
 			 SNDRV_PCM_FMTBIT_S24_LE |\
 			 SNDRV_PCM_FMTBIT_S32_LE )
 
-
 static const struct snd_soc_dai_ops cs42l42_ops = {
+	.startup	= cs42l42_dai_startup,
 	.hw_params	= cs42l42_pcm_hw_params,
 	.set_fmt	= cs42l42_set_dai_fmt,
 	.set_sysclk	= cs42l42_set_sysclk,
-- 
cgit v1.2.3


From e5ada3f6787a4d6234adc6f2f3ae35c6d5b71ba0 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 5 Aug 2021 17:11:07 +0100
Subject: ASoC: cs42l42: Fix mono playback

I2S always has two LRCLK phases and both CH1 and CH2 of the RX
must be enabled (corresponding to the low and high phases of LRCLK.)
The selection of the valid data channels is done by setting the DAC
CHA_SEL and CHB_SEL. CHA_SEL is always the first (left) channel,
CHB_SEL depends on the number of active channels.

Previously for mono ASP CH2 was not enabled, the result was playing
mono data would not produce any audio output.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 621d65f3b868 ("ASoC: cs42l42: Provide finer control on playback path")
Link: https://lore.kernel.org/r/20210805161111.10410-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l42.c | 15 +++++++++++++--
 sound/soc/codecs/cs42l42.h |  2 ++
 2 files changed, 15 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 29e0c8dc8466..99c022be94a6 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -460,8 +460,8 @@ static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HP"),
 	SND_SOC_DAPM_DAC("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1),
 	SND_SOC_DAPM_MIXER("MIXER", CS42L42_PWR_CTL1, CS42L42_MIXER_PDN_SHIFT, 1, NULL, 0),
-	SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH1_SHIFT, 0),
-	SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH2_SHIFT, 0),
+	SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, SND_SOC_NOPM, 0, 0),
 
 	/* Playback Requirements */
 	SND_SOC_DAPM_SUPPLY("ASP DAI0", CS42L42_PWR_CTL1, CS42L42_ASP_DAI_PDN_SHIFT, 1, NULL, 0),
@@ -866,6 +866,17 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
 		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES,
 							 CS42L42_ASP_RX_CH_AP_MASK |
 							 CS42L42_ASP_RX_CH_RES_MASK, val);
+
+		/* Channel B comes from the last active channel */
+		snd_soc_component_update_bits(component, CS42L42_SP_RX_CH_SEL,
+					      CS42L42_SP_RX_CHB_SEL_MASK,
+					      (channels - 1) << CS42L42_SP_RX_CHB_SEL_SHIFT);
+
+		/* Both LRCLK slots must be enabled */
+		snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_EN,
+					      CS42L42_ASP_RX0_CH_EN_MASK,
+					      BIT(CS42L42_ASP_RX0_CH1_SHIFT) |
+					      BIT(CS42L42_ASP_RX0_CH2_SHIFT));
 		break;
 	default:
 		break;
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 3dfeaf6611d1..112ffcf938c3 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -653,6 +653,8 @@
 
 /* Page 0x25 Audio Port Registers */
 #define CS42L42_SP_RX_CH_SEL		(CS42L42_PAGE_25 + 0x01)
+#define CS42L42_SP_RX_CHB_SEL_SHIFT	2
+#define CS42L42_SP_RX_CHB_SEL_MASK	(3 << CS42L42_SP_RX_CHB_SEL_SHIFT)
 
 #define CS42L42_SP_RX_ISOC_CTL		(CS42L42_PAGE_25 + 0x02)
 #define CS42L42_SP_RX_RSYNC_SHIFT	6
-- 
cgit v1.2.3