From 2230c49f09b552454eac51b81e9e4e41060b5e70 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 13 May 2016 16:45:18 +0100
Subject: ASoC: arizona: Add a notifier chain for CODEC events

Add a notifier chain that can be used from the machine driver to catch
events generated by the CODEC.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/arizona.c | 35 +++++++++++++++++++++++++++++++++++
 sound/soc/codecs/arizona.h | 10 ++++++++++
 sound/soc/codecs/cs47l24.c |  1 +
 sound/soc/codecs/wm5110.c  |  1 +
 4 files changed, 47 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 664a8c044ffb..7f9ab92ffa91 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -324,6 +324,17 @@ int arizona_init_gpio(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(arizona_init_gpio);
 
+int arizona_init_notifiers(struct snd_soc_codec *codec)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+
+	BLOCKING_INIT_NOTIFIER_HEAD(&arizona->notifier);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_notifiers);
+
 const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 	"None",
 	"Tone Generator 1",
@@ -2573,6 +2584,30 @@ int arizona_lhpf_coeff_put(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(arizona_lhpf_coeff_put);
 
+int arizona_register_notifier(struct snd_soc_codec *codec,
+			      struct notifier_block *nb,
+			      int (*notify)(struct notifier_block *nb,
+					    unsigned long action, void *data))
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+
+	nb->notifier_call = notify;
+
+	return blocking_notifier_chain_register(&arizona->notifier, nb);
+}
+EXPORT_SYMBOL_GPL(arizona_register_notifier);
+
+int arizona_unregister_notifier(struct snd_soc_codec *codec,
+				struct notifier_block *nb)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+
+	return blocking_notifier_chain_unregister(&arizona->notifier, nb);
+}
+EXPORT_SYMBOL_GPL(arizona_unregister_notifier);
+
 MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ce0531b8c632..245d13c157a5 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -306,6 +306,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source,
 extern int arizona_init_spk(struct snd_soc_codec *codec);
 extern int arizona_init_gpio(struct snd_soc_codec *codec);
 extern int arizona_init_mono(struct snd_soc_codec *codec);
+extern int arizona_init_notifiers(struct snd_soc_codec *codec);
 
 extern int arizona_free_spk(struct snd_soc_codec *codec);
 
@@ -317,4 +318,13 @@ int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
 extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
 
 extern const char *arizona_sample_rate_val_to_name(unsigned int rate_val);
+
+extern int arizona_register_notifier(struct snd_soc_codec *codec,
+				     struct notifier_block *nb,
+				     int (*notify)(struct notifier_block *nb,
+						   unsigned long action,
+						   void *data));
+extern int arizona_unregister_notifier(struct snd_soc_codec *codec,
+				       struct notifier_block *nb);
+
 #endif
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 5ec5a682d186..fa9a6a5a6120 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -1096,6 +1096,7 @@ static int cs47l24_codec_probe(struct snd_soc_codec *codec)
 	arizona_init_spk(codec);
 	arizona_init_gpio(codec);
 	arizona_init_mono(codec);
+	arizona_init_notifiers(codec);
 
 	ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1,
 				  "ADSP2 Compressed IRQ", cs47l24_adsp2_irq,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index b5820e4d5471..338a3b52705b 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2251,6 +2251,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 	arizona_init_spk(codec);
 	arizona_init_gpio(codec);
 	arizona_init_mono(codec);
+	arizona_init_notifiers(codec);
 
 	ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1,
 				  "ADSP2 Compressed IRQ", wm5110_adsp2_irq,
-- 
cgit v1.2.3


From 7baa7e2490e1c292a84406c90089511c96ce3114 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 13 May 2016 16:45:19 +0100
Subject: ASoC: arizona: Add event notification on voice trigger events

Inform the notifier chain if the DSP recognises a voice trigger.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/arizona.h | 3 +++
 sound/soc/codecs/cs47l24.c | 4 ++++
 sound/soc/codecs/wm5110.c  | 4 ++++
 3 files changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 245d13c157a5..18d347f3bfbe 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -63,6 +63,9 @@
 #define ARIZONA_DVFS_SR1_RQ	0x001
 #define ARIZONA_DVFS_ADSP1_RQ	0x100
 
+/* Notifier events */
+#define ARIZONA_NOTIFY_VOICE_TRIGGER   0x1
+
 struct arizona;
 struct wm_adsp;
 
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index fa9a6a5a6120..7e3d138d077b 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -1074,6 +1074,10 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
 		ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
 		if (ret != -ENODEV)
 			serviced++;
+		if (ret == WM_ADSP_COMPR_VOICE_TRIGGER)
+			arizona_call_notifiers(arizona,
+					       ARIZONA_NOTIFY_VOICE_TRIGGER,
+					       (void *)i);
 	}
 
 	if (!serviced) {
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 338a3b52705b..dbc9b4df38a0 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2229,6 +2229,10 @@ static irqreturn_t wm5110_adsp2_irq(int irq, void *data)
 		ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
 		if (ret != -ENODEV)
 			serviced++;
+		if (ret == WM_ADSP_COMPR_VOICE_TRIGGER)
+			arizona_call_notifiers(arizona,
+					       ARIZONA_NOTIFY_VOICE_TRIGGER,
+					       (void *)i);
 	}
 
 	if (!serviced) {
-- 
cgit v1.2.3


From 546ad3d024fd47e5042d80ae1dc4c7d1b00912a7 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Tue, 31 May 2016 12:44:17 +0100
Subject: ASoC: arizona: Add data structure for voice trigger notifier

64-bit builds would generate a warning when we passed the core number as
a pointer through the notifier data:

sound/soc/codecs/cs47l24.c:1091:13: warning: cast to pointer from
integer of different size [-Wint-to-pointer-cast]
             (void *)i);

Rather than just fix this up with more casting add a data structure that
holds information for the notifier chain. This will make it easier to
add additional information in the future as well.

Fixes: 7baa7e2490e1 ("ASoC: arizona: Add event notification on voice trigger events")
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/arizona.h | 4 ++++
 sound/soc/codecs/cs47l24.c | 7 +++++--
 sound/soc/codecs/wm5110.c  | 7 +++++--
 3 files changed, 14 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 18d347f3bfbe..46862af7665e 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -98,6 +98,10 @@ struct arizona_priv {
 	bool dvfs_cached;
 };
 
+struct arizona_voice_trigger_info {
+	int core;
+};
+
 #define ARIZONA_NUM_MIXER_INPUTS 104
 
 extern const unsigned int arizona_mixer_tlv[];
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 7e3d138d077b..bbc8cf18ded0 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -1067,6 +1067,7 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
 {
 	struct cs47l24_priv *priv = data;
 	struct arizona *arizona = priv->core.arizona;
+	struct arizona_voice_trigger_info info;
 	int serviced = 0;
 	int i, ret;
 
@@ -1074,10 +1075,12 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
 		ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
 		if (ret != -ENODEV)
 			serviced++;
-		if (ret == WM_ADSP_COMPR_VOICE_TRIGGER)
+		if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) {
+			info.core = i;
 			arizona_call_notifiers(arizona,
 					       ARIZONA_NOTIFY_VOICE_TRIGGER,
-					       (void *)i);
+					       &info);
+		}
 	}
 
 	if (!serviced) {
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index dbc9b4df38a0..83c48eca56f5 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2222,6 +2222,7 @@ static irqreturn_t wm5110_adsp2_irq(int irq, void *data)
 {
 	struct wm5110_priv *priv = data;
 	struct arizona *arizona = priv->core.arizona;
+	struct arizona_voice_trigger_info info;
 	int serviced = 0;
 	int i, ret;
 
@@ -2229,10 +2230,12 @@ static irqreturn_t wm5110_adsp2_irq(int irq, void *data)
 		ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
 		if (ret != -ENODEV)
 			serviced++;
-		if (ret == WM_ADSP_COMPR_VOICE_TRIGGER)
+		if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) {
+			info.core = i;
 			arizona_call_notifiers(arizona,
 					       ARIZONA_NOTIFY_VOICE_TRIGGER,
-					       (void *)i);
+					       &info);
+		}
 	}
 
 	if (!serviced) {
-- 
cgit v1.2.3


From a2ebd58627e9aa486fccbbdda6338675bdf3e267 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Tue, 31 May 2016 13:34:59 +0300
Subject: ASoC: ak4642: Implement suspend callback

Add the suspend callback to accompany the existing resume operation.
With the suspend/resume callbacks the regmap (regcache) state handling can
follow the recommended sequence.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ak4642.c | 13 +++++++++++--
 1 file changed, 11 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4d8b9e49e8d6..cc941d66ec3d 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -523,15 +523,23 @@ static struct snd_soc_dai_driver ak4642_dai = {
 	.symmetric_rates = 1,
 };
 
-static int ak4642_resume(struct snd_soc_codec *codec)
+static int ak4642_suspend(struct snd_soc_codec *codec)
 {
 	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
 
+	regcache_cache_only(regmap, true);
 	regcache_mark_dirty(regmap);
-	regcache_sync(regmap);
 	return 0;
 }
 
+static int ak4642_resume(struct snd_soc_codec *codec)
+{
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+	regcache_cache_only(regmap, false);
+	regcache_sync(regmap);
+	return 0;
+}
 static int ak4642_probe(struct snd_soc_codec *codec)
 {
 	struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
@@ -544,6 +552,7 @@ static int ak4642_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
 	.probe			= ak4642_probe,
+	.suspend		= ak4642_suspend,
 	.resume			= ak4642_resume,
 	.set_bias_level		= ak4642_set_bias_level,
 	.controls		= ak4642_snd_controls,
-- 
cgit v1.2.3


From a0d3546cf9e5123fca1468651ca99d469d202198 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 2 Jun 2016 14:19:42 +0200
Subject: ASoC: Add ADAU7002 Stereo PDM-to-I2S/TDM Converter driver

This patch adds support for the ADAU7002 PDM-to-I2S/TDM converter. The
ADAU7002 takes a stereo PDM signal (e.g. from two digital microphones) and
converts it into a I2S/TDM PCM stream. The chip does not have a
programmable control interface and the driver simply describes the static
capabilities of the chip.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig    |  4 +++
 sound/soc/codecs/Makefile   |  2 ++
 sound/soc/codecs/adau7002.c | 80 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 86 insertions(+)
 create mode 100644 sound/soc/codecs/adau7002.c

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4d82a58ff6b0..9ae3c0a2077a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ADAU1977_SPI if SPI_MASTER
 	select SND_SOC_ADAU1977_I2C if I2C
 	select SND_SOC_ADAU1701 if I2C
+	select SND_SOC_ADAU7002
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
@@ -322,6 +323,9 @@ config SND_SOC_ADAU1977_I2C
 	select SND_SOC_ADAU1977
 	select REGMAP_I2C
 
+config SND_SOC_ADAU7002
+	tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter"
+
 config SND_SOC_ADAV80X
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 0f548fd34ca3..9f8167668e06 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -19,6 +19,7 @@ snd-soc-adau1781-spi-objs := adau1781-spi.o
 snd-soc-adau1977-objs := adau1977.o
 snd-soc-adau1977-spi-objs := adau1977-spi.o
 snd-soc-adau1977-i2c-objs := adau1977-i2c.o
+snd-soc-adau7002-objs := adau7002.o
 snd-soc-adav80x-objs := adav80x.o
 snd-soc-adav801-objs := adav801.o
 snd-soc-adav803-objs := adav803.o
@@ -232,6 +233,7 @@ obj-$(CONFIG_SND_SOC_ADAU1781_SPI)	+= snd-soc-adau1781-spi.o
 obj-$(CONFIG_SND_SOC_ADAU1977)		+= snd-soc-adau1977.o
 obj-$(CONFIG_SND_SOC_ADAU1977_SPI)	+= snd-soc-adau1977-spi.o
 obj-$(CONFIG_SND_SOC_ADAU1977_I2C)	+= snd-soc-adau1977-i2c.o
+obj-$(CONFIG_SND_SOC_ADAU7002)	+= snd-soc-adau7002.o
 obj-$(CONFIG_SND_SOC_ADAV80X)  += snd-soc-adav80x.o
 obj-$(CONFIG_SND_SOC_ADAV801)  += snd-soc-adav801.o
 obj-$(CONFIG_SND_SOC_ADAV803)  += snd-soc-adav803.o
diff --git a/sound/soc/codecs/adau7002.c b/sound/soc/codecs/adau7002.c
new file mode 100644
index 000000000000..9df72c6adcca
--- /dev/null
+++ b/sound/soc/codecs/adau7002.c
@@ -0,0 +1,80 @@
+/*
+ * ADAU7002 Stereo PDM-to-I2S/TDM converter driver
+ *
+ * Copyright 2014-2016 Analog Devices
+ *  Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static const struct snd_soc_dapm_widget adau7002_widgets[] = {
+	SND_SOC_DAPM_INPUT("PDM_DAT"),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("IOVDD", 0, 0),
+};
+
+static const struct snd_soc_dapm_route adau7002_routes[] = {
+	{ "Capture", NULL, "PDM_DAT" },
+	{ "Capture", NULL, "IOVDD" },
+};
+
+static struct snd_soc_dai_driver adau7002_dai = {
+	.name = "adau7002-hifi",
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE,
+		.sig_bits = 20,
+	},
+};
+
+static const struct snd_soc_codec_driver adau7002_codec_driver = {
+	.dapm_widgets = adau7002_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(adau7002_widgets),
+	.dapm_routes = adau7002_routes,
+	.num_dapm_routes = ARRAY_SIZE(adau7002_routes),
+};
+
+static int adau7002_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &adau7002_codec_driver,
+			&adau7002_dai, 1);
+}
+
+static int adau7002_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id adau7002_dt_ids[] = {
+	{ .compatible = "adi,adau7002", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, adau7002_dt_ids);
+#endif
+
+static struct platform_driver adau7002_driver = {
+	.driver = {
+		.name = "adau7002",
+		.of_match_table	= of_match_ptr(adau7002_dt_ids),
+	},
+	.probe = adau7002_probe,
+	.remove = adau7002_remove,
+};
+module_platform_driver(adau7002_driver);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ADAU7002 Stereo PDM-to-I2S/TDM Converter driver");
+MODULE_LICENSE("GPL v2");
-- 
cgit v1.2.3


From 0eadaa9ce2aacdcc3cf050d98c25aacabadc557f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 9 Jun 2016 19:39:06 +0200
Subject: ASoC: adau: Factor out shared PLL configuration code

Multiple devices from the ADAU family share the same PLL structure and
configuration register layout. Introduce a new helper module that can be
used to calculated the PLL configuration registers based on a specified
input frequency and the desired output frequency of the PLL.

The ADAU1761/ADAU1781 and ADAU1373 drivers are updated to make use of this
new helper module. But future drivers for additional devices from the ADAU
family are also expected to make use of it.

In anticipation of sharing more infrastructure code between different
devices from the ADAU family the new module is called adau-utils.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig      |  5 ++++
 sound/soc/codecs/Makefile     |  2 ++
 sound/soc/codecs/adau-utils.c | 61 +++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/adau-utils.h |  7 +++++
 sound/soc/codecs/adau1373.c   | 38 +++++++--------------------
 sound/soc/codecs/adau17x1.c   | 38 +++------------------------
 6 files changed, 89 insertions(+), 62 deletions(-)
 create mode 100644 sound/soc/codecs/adau-utils.c
 create mode 100644 sound/soc/codecs/adau-utils.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4d82a58ff6b0..3e215395a199 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -269,8 +269,12 @@ config SND_SOC_AD1980
 config SND_SOC_AD73311
 	tristate
 
+config SND_SOC_ADAU_UTILS
+	tristate
+
 config SND_SOC_ADAU1373
 	tristate
+	select SND_SOC_ADAU_UTILS
 
 config SND_SOC_ADAU1701
 	tristate "Analog Devices ADAU1701 CODEC"
@@ -280,6 +284,7 @@ config SND_SOC_ADAU1701
 config SND_SOC_ADAU17X1
 	tristate
 	select SND_SOC_SIGMADSP_REGMAP
+	select SND_SOC_ADAU_UTILS
 
 config SND_SOC_ADAU1761
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 0f548fd34ca3..d61957f2618c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -7,6 +7,7 @@ snd-soc-ad193x-spi-objs := ad193x-spi.o
 snd-soc-ad193x-i2c-objs := ad193x-i2c.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
+snd-soc-adau-utils-objs := adau-utils.o
 snd-soc-adau1373-objs := adau1373.o
 snd-soc-adau1701-objs := adau1701.o
 snd-soc-adau17x1-objs := adau17x1.o
@@ -220,6 +221,7 @@ obj-$(CONFIG_SND_SOC_AD193X_SPI)	+= snd-soc-ad193x-spi.o
 obj-$(CONFIG_SND_SOC_AD193X_I2C)	+= snd-soc-ad193x-i2c.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU_UTILS)	+= snd-soc-adau-utils.o
 obj-$(CONFIG_SND_SOC_ADAU1373)	+= snd-soc-adau1373.o
 obj-$(CONFIG_SND_SOC_ADAU1701)		+= snd-soc-adau1701.o
 obj-$(CONFIG_SND_SOC_ADAU17X1)		+= snd-soc-adau17x1.o
diff --git a/sound/soc/codecs/adau-utils.c b/sound/soc/codecs/adau-utils.c
new file mode 100644
index 000000000000..19d6a6f41b12
--- /dev/null
+++ b/sound/soc/codecs/adau-utils.c
@@ -0,0 +1,61 @@
+/*
+ * Shared helper functions for devices from the ADAU family
+ *
+ * Copyright 2011-2016 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/gcd.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+
+#include "adau-utils.h"
+
+int adau_calc_pll_cfg(unsigned int freq_in, unsigned int freq_out,
+	uint8_t regs[5])
+{
+	unsigned int r, n, m, i, j;
+	unsigned int div;
+
+	if (!freq_out) {
+		r = 0;
+		n = 0;
+		m = 0;
+		div = 0;
+	} else {
+		if (freq_out % freq_in != 0) {
+			div = DIV_ROUND_UP(freq_in, 13500000);
+			freq_in /= div;
+			r = freq_out / freq_in;
+			i = freq_out % freq_in;
+			j = gcd(i, freq_in);
+			n = i / j;
+			m = freq_in / j;
+			div--;
+		} else {
+			r = freq_out / freq_in;
+			n = 0;
+			m = 0;
+			div = 0;
+		}
+		if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2)
+			return -EINVAL;
+	}
+
+	regs[0] = m >> 8;
+	regs[1] = m & 0xff;
+	regs[2] = n >> 8;
+	regs[3] = n & 0xff;
+	regs[4] = (r << 3) | (div << 1);
+	if (m != 0)
+		regs[4] |= 1; /* Fractional mode */
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(adau_calc_pll_cfg);
+
+MODULE_DESCRIPTION("ASoC ADAU audio CODECs shared helper functions");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/adau-utils.h b/sound/soc/codecs/adau-utils.h
new file mode 100644
index 000000000000..939b5f37762f
--- /dev/null
+++ b/sound/soc/codecs/adau-utils.h
@@ -0,0 +1,7 @@
+#ifndef SOUND_SOC_CODECS_ADAU_PLL_H
+#define SOUND_SOC_CODECS_ADAU_PLL_H
+
+int adau_calc_pll_cfg(unsigned int freq_in, unsigned int freq_out,
+	uint8_t regs[5]);
+
+#endif
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index fe1353a797b9..1556b360fa15 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -23,6 +23,7 @@
 #include <sound/adau1373.h>
 
 #include "adau1373.h"
+#include "adau-utils.h"
 
 struct adau1373_dai {
 	unsigned int clk_src;
@@ -1254,7 +1255,8 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
 {
 	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
 	unsigned int dpll_div = 0;
-	unsigned int x, r, n, m, i, j, mode;
+	uint8_t pll_regs[5];
+	int ret;
 
 	switch (pll_id) {
 	case ADAU1373_PLL1:
@@ -1295,27 +1297,8 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
 		dpll_div++;
 	}
 
-	if (freq_out % freq_in != 0) {
-		/* fout = fin * (r + (n/m)) / x */
-		x = DIV_ROUND_UP(freq_in, 13500000);
-		freq_in /= x;
-		r = freq_out / freq_in;
-		i = freq_out % freq_in;
-		j = gcd(i, freq_in);
-		n = i / j;
-		m = freq_in / j;
-		x--;
-		mode = 1;
-	} else {
-		/* fout = fin / r */
-		r = freq_out / freq_in;
-		n = 0;
-		m = 0;
-		x = 0;
-		mode = 0;
-	}
-
-	if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff)
+	ret = adau_calc_pll_cfg(freq_in, freq_out, pll_regs);
+	if (ret)
 		return -EINVAL;
 
 	if (dpll_div) {
@@ -1330,12 +1313,11 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
 
 	regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id),
 		(source << 4) | dpll_div);
-	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff);
-	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff);
-	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff);
-	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff);
-	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id),
-		(r << 3) | (x << 1) | mode);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), pll_regs[0]);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), pll_regs[1]);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), pll_regs[2]);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), pll_regs[3]);
+	regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), pll_regs[4]);
 
 	/* Set sysclk to pll_rate / 4 */
 	regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09);
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index fcf05b254ecd..66a6e061923d 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -23,6 +23,7 @@
 
 #include "sigmadsp.h"
 #include "adau17x1.h"
+#include "adau-utils.h"
 
 static const char * const adau17x1_capture_mixer_boost_text[] = {
 	"Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3",
@@ -391,45 +392,14 @@ static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
-	unsigned int r, n, m, i, j;
-	unsigned int div;
 	int ret;
 
 	if (freq_in < 8000000 || freq_in > 27000000)
 		return -EINVAL;
 
-	if (!freq_out) {
-		r = 0;
-		n = 0;
-		m = 0;
-		div = 0;
-	} else {
-		if (freq_out % freq_in != 0) {
-			div = DIV_ROUND_UP(freq_in, 13500000);
-			freq_in /= div;
-			r = freq_out / freq_in;
-			i = freq_out % freq_in;
-			j = gcd(i, freq_in);
-			n = i / j;
-			m = freq_in / j;
-			div--;
-		} else {
-			r = freq_out / freq_in;
-			n = 0;
-			m = 0;
-			div = 0;
-		}
-		if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2)
-			return -EINVAL;
-	}
-
-	adau->pll_regs[0] = m >> 8;
-	adau->pll_regs[1] = m & 0xff;
-	adau->pll_regs[2] = n >> 8;
-	adau->pll_regs[3] = n & 0xff;
-	adau->pll_regs[4] = (r << 3) | (div << 1);
-	if (m != 0)
-		adau->pll_regs[4] |= 1; /* Fractional mode */
+	ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs);
+	if (ret < 0)
+		return ret;
 
 	/* The PLL register is 6 bytes long and can only be written at once. */
 	ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL,
-- 
cgit v1.2.3


From a4f2d87c63571d4cd9467d369f2fbf2362646043 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Mon, 13 Jun 2016 14:17:10 +0100
Subject: ALSA: compress: Add function to indicate the stream has gone bad

Currently, the avail IOCTL doesn't pass any error status, which
means typically on error it simply shows no data available. This
can lead to situations where user-space is waiting indefinitely
for data that will never come as the DSP has suffered an
unrecoverable error.

Add snd_compr_stop_error which end drivers can call to indicate
the stream has suffered an unrecoverable error and stop it. The
avail and poll IOCTLs are then updated to report if the stream is
in an error state to user-space. Allowing the error to propagate
out. Processing of the actual snd_compr_stop needs to be deferred
to a worker thread as the end driver may detect the errors during
an existing operation callback.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/core/compress_offload.c | 67 +++++++++++++++++++++++++++++++++++++++++--
 1 file changed, 65 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 9b3334be9df2..2c498488af6c 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -67,6 +67,8 @@ struct snd_compr_file {
 	struct snd_compr_stream stream;
 };
 
+static void error_delayed_work(struct work_struct *work);
+
 /*
  * a note on stream states used:
  * we use following states in the compressed core
@@ -123,6 +125,9 @@ static int snd_compr_open(struct inode *inode, struct file *f)
 		snd_card_unref(compr->card);
 		return -ENOMEM;
 	}
+
+	INIT_DELAYED_WORK(&data->stream.error_work, error_delayed_work);
+
 	data->stream.ops = compr->ops;
 	data->stream.direction = dirn;
 	data->stream.private_data = compr->private_data;
@@ -153,6 +158,8 @@ static int snd_compr_free(struct inode *inode, struct file *f)
 	struct snd_compr_file *data = f->private_data;
 	struct snd_compr_runtime *runtime = data->stream.runtime;
 
+	cancel_delayed_work_sync(&data->stream.error_work);
+
 	switch (runtime->state) {
 	case SNDRV_PCM_STATE_RUNNING:
 	case SNDRV_PCM_STATE_DRAINING:
@@ -237,6 +244,15 @@ snd_compr_ioctl_avail(struct snd_compr_stream *stream, unsigned long arg)
 	avail = snd_compr_calc_avail(stream, &ioctl_avail);
 	ioctl_avail.avail = avail;
 
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+		return -EBADFD;
+	case SNDRV_PCM_STATE_XRUN:
+		return -EPIPE;
+	default:
+		break;
+	}
+
 	if (copy_to_user((__u64 __user *)arg,
 				&ioctl_avail, sizeof(ioctl_avail)))
 		return -EFAULT;
@@ -346,11 +362,13 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf,
 	switch (stream->runtime->state) {
 	case SNDRV_PCM_STATE_OPEN:
 	case SNDRV_PCM_STATE_PREPARED:
-	case SNDRV_PCM_STATE_XRUN:
 	case SNDRV_PCM_STATE_SUSPENDED:
 	case SNDRV_PCM_STATE_DISCONNECTED:
 		retval = -EBADFD;
 		goto out;
+	case SNDRV_PCM_STATE_XRUN:
+		retval = -EPIPE;
+		goto out;
 	}
 
 	avail = snd_compr_get_avail(stream);
@@ -399,10 +417,16 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait)
 	stream = &data->stream;
 
 	mutex_lock(&stream->device->lock);
-	if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) {
+
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_XRUN:
 		retval = snd_compr_get_poll(stream) | POLLERR;
 		goto out;
+	default:
+		break;
 	}
+
 	poll_wait(f, &stream->runtime->sleep, wait);
 
 	avail = snd_compr_get_avail(stream);
@@ -697,6 +721,45 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
 	return retval;
 }
 
+static void error_delayed_work(struct work_struct *work)
+{
+	struct snd_compr_stream *stream;
+
+	stream = container_of(work, struct snd_compr_stream, error_work.work);
+
+	mutex_lock(&stream->device->lock);
+
+	stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
+	wake_up(&stream->runtime->sleep);
+
+	mutex_unlock(&stream->device->lock);
+}
+
+/*
+ * snd_compr_stop_error: Report a fatal error on a stream
+ * @stream: pointer to stream
+ * @state: state to transition the stream to
+ *
+ * Stop the stream and set its state.
+ *
+ * Should be called with compressed device lock held.
+ */
+int snd_compr_stop_error(struct snd_compr_stream *stream,
+			 snd_pcm_state_t state)
+{
+	if (stream->runtime->state == state)
+		return 0;
+
+	stream->runtime->state = state;
+
+	pr_debug("Changing state to: %d\n", state);
+
+	queue_delayed_work(system_power_efficient_wq, &stream->error_work, 0);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_compr_stop_error);
+
 static int snd_compress_wait_for_drain(struct snd_compr_stream *stream)
 {
 	int ret;
-- 
cgit v1.2.3


From 8d280664d26538cd37e7c08b1c2b58fe006cc482 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Mon, 13 Jun 2016 14:17:11 +0100
Subject: ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure

If we encounter a fatal error on the compressed stream call the new
snd_compr_stop_error to shutdown the stream and allow the core to
inform user-space that the stream is no longer valid.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm_adsp.c | 11 +++++++++--
 1 file changed, 9 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index a07bd7c2c587..8ed1cdececf2 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -3043,6 +3043,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
 	}
 
 	if (compr->buf->error) {
+		snd_compr_stop_error(stream, SNDRV_PCM_STATE_XRUN);
 		ret = -EIO;
 		goto out;
 	}
@@ -3060,8 +3061,12 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
 		 */
 		if (buf->avail < wm_adsp_compr_frag_words(compr)) {
 			ret = wm_adsp_buffer_get_error(buf);
-			if (ret < 0)
+			if (ret < 0) {
+				if (compr->buf->error)
+					snd_compr_stop_error(stream,
+							SNDRV_PCM_STATE_XRUN);
 				goto out;
+			}
 
 			ret = wm_adsp_buffer_reenable_irq(buf);
 			if (ret < 0) {
@@ -3159,8 +3164,10 @@ static int wm_adsp_compr_read(struct wm_adsp_compr *compr,
 	if (!compr->buf)
 		return -ENXIO;
 
-	if (compr->buf->error)
+	if (compr->buf->error) {
+		snd_compr_stop_error(compr->stream, SNDRV_PCM_STATE_XRUN);
 		return -EIO;
+	}
 
 	count /= WM_ADSP_DATA_WORD_SIZE;
 
-- 
cgit v1.2.3


From 28ee3d73773e2d9ae922f7496723ab5c92cc16de Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Mon, 13 Jun 2016 14:17:12 +0100
Subject: ASoC: wm_adsp: Treat missing compressed buffer as a fatal error

If the DSP is powered down whilst a compressed stream is being processed
we should treat this as a fatal error, clearly the stream is no longer
valid.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm_adsp.c | 12 ++----------
 1 file changed, 2 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 8ed1cdececf2..7e42474d7ae4 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -3037,12 +3037,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
 
 	buf = compr->buf;
 
-	if (!compr->buf) {
-		ret = -ENXIO;
-		goto out;
-	}
-
-	if (compr->buf->error) {
+	if (!compr->buf || compr->buf->error) {
 		snd_compr_stop_error(stream, SNDRV_PCM_STATE_XRUN);
 		ret = -EIO;
 		goto out;
@@ -3161,10 +3156,7 @@ static int wm_adsp_compr_read(struct wm_adsp_compr *compr,
 
 	adsp_dbg(dsp, "Requested read of %zu bytes\n", count);
 
-	if (!compr->buf)
-		return -ENXIO;
-
-	if (compr->buf->error) {
+	if (!compr->buf || compr->buf->error) {
 		snd_compr_stop_error(compr->stream, SNDRV_PCM_STATE_XRUN);
 		return -EIO;
 	}
-- 
cgit v1.2.3


From 5d76de61dd8cb89b7189ef7456fba921c547c398 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 15 Jun 2016 15:07:27 +0200
Subject: ASoC: adau17x1: Add support for specifying the MCLK using the CCF

The devices from the ADAU17X1 family all have a MCLK clock input which
supplies the master clock for the device. The master clock is used as the
input clock for the PLL. Currently the MCLK rate as well as the desired PLL
output frequency need to be supplied by calling snd_soc_dai_set_pll() form
a machine driver.

Add support for specifying the MCLK using the common clock framework. In
addition to that also automatically configure the PLL to a suitable rate
if the master clock was provided using the CCW. This allows to use the
CODEC driver without any special configuration requirements from the
machine driver.

While the PLL output frequency can be configured over a (more or less)
continuous range the narrowness of the range and the other constraints of
the clocking tree usually only result in two output frequencies that will
actually be chosen. One for 44.1kHz based rates and one for 48kHz based
rates, these are the rates that the automatic PLL configuration will use.
For the rare case where a non-standard setup is required a machine driver
can disable the auto-configuration and configure a custom frequency using
the existing mechanisms.

If the common clock framework is not enabled clk_get() will return NULL and
the driver will function as before and the clock rate needs to be
configured manually.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/adau1761-i2c.c |   2 +-
 sound/soc/codecs/adau1761-spi.c |   2 +-
 sound/soc/codecs/adau1781-i2c.c |   2 +-
 sound/soc/codecs/adau1781-spi.c |   2 +-
 sound/soc/codecs/adau17x1.c     | 219 +++++++++++++++++++++++++++++-----------
 sound/soc/codecs/adau17x1.h     |   6 ++
 6 files changed, 171 insertions(+), 62 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c
index 8de010f758cd..9e7f257f17f8 100644
--- a/sound/soc/codecs/adau1761-i2c.c
+++ b/sound/soc/codecs/adau1761-i2c.c
@@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client,
 
 static int adau1761_i2c_remove(struct i2c_client *client)
 {
-	snd_soc_unregister_codec(&client->dev);
+	adau17x1_remove(&client->dev);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c
index d9171245bd9f..a0b214be759a 100644
--- a/sound/soc/codecs/adau1761-spi.c
+++ b/sound/soc/codecs/adau1761-spi.c
@@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi)
 
 static int adau1761_spi_remove(struct spi_device *spi)
 {
-	snd_soc_unregister_codec(&spi->dev);
+	adau17x1_remove(&spi->dev);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c
index 06cbca84cf02..7b9d1802d159 100644
--- a/sound/soc/codecs/adau1781-i2c.c
+++ b/sound/soc/codecs/adau1781-i2c.c
@@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client,
 
 static int adau1781_i2c_remove(struct i2c_client *client)
 {
-	snd_soc_unregister_codec(&client->dev);
+	adau17x1_remove(&client->dev);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c
index 3d965a01b99c..9b233544d2e8 100644
--- a/sound/soc/codecs/adau1781-spi.c
+++ b/sound/soc/codecs/adau1781-spi.c
@@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi)
 
 static int adau1781_spi_remove(struct spi_device *spi)
 {
-	snd_soc_unregister_codec(&spi->dev);
+	adau17x1_remove(&spi->dev);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 66a6e061923d..439aa3ff1f99 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -9,6 +9,7 @@
 
 #include <linux/module.h>
 #include <linux/init.h>
+#include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -303,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau)
 }
 EXPORT_SYMBOL_GPL(adau17x1_has_dsp);
 
+static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
+	int source, unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adau *adau = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	if (freq_in < 8000000 || freq_in > 27000000)
+		return -EINVAL;
+
+	ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs);
+	if (ret < 0)
+		return ret;
+
+	/* The PLL register is 6 bytes long and can only be written at once. */
+	ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL,
+			adau->pll_regs, ARRAY_SIZE(adau->pll_regs));
+	if (ret)
+		return ret;
+
+	adau->pll_freq = freq_out;
+
+	return 0;
+}
+
+static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec);
+	struct adau *adau = snd_soc_codec_get_drvdata(dai->codec);
+	bool is_pll;
+	bool was_pll;
+
+	switch (clk_id) {
+	case ADAU17X1_CLK_SRC_MCLK:
+		is_pll = false;
+		break;
+	case ADAU17X1_CLK_SRC_PLL_AUTO:
+		if (!adau->mclk)
+			return -EINVAL;
+		/* Fall-through */
+	case ADAU17X1_CLK_SRC_PLL:
+		is_pll = true;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (adau->clk_src) {
+	case ADAU17X1_CLK_SRC_MCLK:
+		was_pll = false;
+		break;
+	case ADAU17X1_CLK_SRC_PLL:
+	case ADAU17X1_CLK_SRC_PLL_AUTO:
+		was_pll = true;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	adau->sysclk = freq;
+
+	if (is_pll != was_pll) {
+		if (is_pll) {
+			snd_soc_dapm_add_routes(dapm,
+				&adau17x1_dapm_pll_route, 1);
+		} else {
+			snd_soc_dapm_del_routes(dapm,
+				&adau17x1_dapm_pll_route, 1);
+		}
+	}
+
+	adau->clk_src = clk_id;
+
+	return 0;
+}
+
+static int adau17x1_auto_pll(struct snd_soc_dai *dai,
+	struct snd_pcm_hw_params *params)
+{
+	struct adau *adau = snd_soc_dai_get_drvdata(dai);
+	unsigned int pll_rate;
+
+	switch (params_rate(params)) {
+	case 48000:
+	case 8000:
+	case 12000:
+	case 16000:
+	case 24000:
+	case 32000:
+	case 96000:
+		pll_rate = 48000 * 1024;
+		break;
+	case 44100:
+	case 7350:
+	case 11025:
+	case 14700:
+	case 22050:
+	case 29400:
+	case 88200:
+		pll_rate = 44100 * 1024;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK,
+		clk_get_rate(adau->mclk), pll_rate);
+}
+
 static int adau17x1_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
@@ -312,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
 	unsigned int freq;
 	int ret;
 
-	if (adau->clk_src == ADAU17X1_CLK_SRC_PLL)
+	switch (adau->clk_src) {
+	case ADAU17X1_CLK_SRC_PLL_AUTO:
+		ret = adau17x1_auto_pll(dai, params);
+		if (ret)
+			return ret;
+		/* Fall-through */
+	case ADAU17X1_CLK_SRC_PLL:
 		freq = adau->pll_freq;
-	else
+		break;
+	default:
 		freq = adau->sysclk;
+		break;
+	}
 
 	if (freq % params_rate(params) != 0)
 		return -EINVAL;
@@ -387,62 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
 			ADAU17X1_SERIAL_PORT1_DELAY_MASK, val);
 }
 
-static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
-	int source, unsigned int freq_in, unsigned int freq_out)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	struct adau *adau = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	if (freq_in < 8000000 || freq_in > 27000000)
-		return -EINVAL;
-
-	ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs);
-	if (ret < 0)
-		return ret;
-
-	/* The PLL register is 6 bytes long and can only be written at once. */
-	ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL,
-			adau->pll_regs, ARRAY_SIZE(adau->pll_regs));
-	if (ret)
-		return ret;
-
-	adau->pll_freq = freq_out;
-
-	return 0;
-}
-
-static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai,
-		int clk_id, unsigned int freq, int dir)
-{
-	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec);
-	struct adau *adau = snd_soc_codec_get_drvdata(dai->codec);
-
-	switch (clk_id) {
-	case ADAU17X1_CLK_SRC_MCLK:
-	case ADAU17X1_CLK_SRC_PLL:
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	adau->sysclk = freq;
-
-	if (adau->clk_src != clk_id) {
-		if (clk_id == ADAU17X1_CLK_SRC_PLL) {
-			snd_soc_dapm_add_routes(dapm,
-				&adau17x1_dapm_pll_route, 1);
-		} else {
-			snd_soc_dapm_del_routes(dapm,
-				&adau17x1_dapm_pll_route, 1);
-		}
-	}
-
-	adau->clk_src = clk_id;
-
-	return 0;
-}
-
 static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai,
 		unsigned int fmt)
 {
@@ -827,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec)
 		ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes,
 			ARRAY_SIZE(adau17x1_no_dsp_dapm_routes));
 	}
+
+	if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK)
+		snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1);
+
 	return ret;
 }
 EXPORT_SYMBOL_GPL(adau17x1_add_routes);
@@ -849,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	const char *firmware_name)
 {
 	struct adau *adau;
+	int ret;
 
 	if (IS_ERR(regmap))
 		return PTR_ERR(regmap);
@@ -857,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	if (!adau)
 		return -ENOMEM;
 
+	adau->mclk = devm_clk_get(dev, "mclk");
+	if (IS_ERR(adau->mclk)) {
+		if (PTR_ERR(adau->mclk) != -ENOENT)
+			return PTR_ERR(adau->mclk);
+		/* Clock is optional (for the driver) */
+		adau->mclk = NULL;
+	} else if (adau->mclk) {
+		adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO;
+
+		/*
+		 * Any valid PLL output rate will work at this point, use one
+		 * that is likely to be chosen later as well. The register will
+		 * be written when the PLL is powered up for the first time.
+		 */
+		ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024,
+				adau->pll_regs);
+		if (ret < 0)
+			return ret;
+
+		ret = clk_prepare_enable(adau->mclk);
+		if (ret)
+			return ret;
+	}
+
 	adau->regmap = regmap;
 	adau->switch_mode = switch_mode;
 	adau->type = type;
@@ -880,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
 }
 EXPORT_SYMBOL_GPL(adau17x1_probe);
 
+void adau17x1_remove(struct device *dev)
+{
+	struct adau *adau = dev_get_drvdata(dev);
+
+	snd_soc_unregister_codec(dev);
+	if (adau->mclk)
+		clk_disable_unprepare(adau->mclk);
+}
+EXPORT_SYMBOL_GPL(adau17x1_remove);
+
 MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code");
 MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index 5ae87a084d97..bf04b7efee40 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -22,13 +22,18 @@ enum adau17x1_pll_src {
 };
 
 enum adau17x1_clk_src {
+	/* Automatically configure PLL based on the sample rate */
+	ADAU17X1_CLK_SRC_PLL_AUTO,
 	ADAU17X1_CLK_SRC_MCLK,
 	ADAU17X1_CLK_SRC_PLL,
 };
 
+struct clk;
+
 struct adau {
 	unsigned int sysclk;
 	unsigned int pll_freq;
+	struct clk *mclk;
 
 	enum adau17x1_clk_src clk_src;
 	enum adau17x1_type type;
@@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec);
 int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	enum adau17x1_type type, void (*switch_mode)(struct device *dev),
 	const char *firmware_name);
+void adau17x1_remove(struct device *dev);
 int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec,
 	enum adau17x1_micbias_voltage micbias);
 bool adau17x1_readable_register(struct device *dev, unsigned int reg);
-- 
cgit v1.2.3


From f9ae17ba97e0fd134f7c0108c70e708313a07063 Mon Sep 17 00:00:00 2001
From: Geert Uytterhoeven <geert+renesas@glider.be>
Date: Thu, 16 Jun 2016 14:34:31 +0200
Subject: ASoC: ak4613: Implement suspend callback

Add the suspend callback to accompany the existing resume operation.
With the suspend/resume callbacks the regmap (regcache) state handling
can follow the recommended sequence.

Based on commit a2ebd58627e9aa48 ("ASoC: ak4642: Implement suspend
callback") by Peter Ujfalusi <peter.ujfalusi@ti.com>.

Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ak4613.c | 12 +++++++++++-
 1 file changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 5013d2ba0c10..97798d250f08 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -437,15 +437,25 @@ static struct snd_soc_dai_driver ak4613_dai = {
 	.symmetric_rates = 1,
 };
 
-static int ak4613_resume(struct snd_soc_codec *codec)
+static int ak4613_suspend(struct snd_soc_codec *codec)
 {
 	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
 
+	regcache_cache_only(regmap, true);
 	regcache_mark_dirty(regmap);
+	return 0;
+}
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+	regcache_cache_only(regmap, false);
 	return regcache_sync(regmap);
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+	.suspend		= ak4613_suspend,
 	.resume			= ak4613_resume,
 	.set_bias_level		= ak4613_set_bias_level,
 	.controls		= ak4613_snd_controls,
-- 
cgit v1.2.3


From 6facd2d10f828d14dd7a38153cd7814d92a47397 Mon Sep 17 00:00:00 2001
From: Simon Trimmer <simont@opensource.wolfsonmicro.com>
Date: Wed, 22 Jun 2016 15:31:03 +0100
Subject: ASoC: wm_adsp: Disable DMAs before clearing the transfer length

This patch reorders the clearing of the DMA masks to avoid potential
artefacts being introduced.

Signed-off-by: Simon Trimmer <simont@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm_adsp.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 7e42474d7ae4..f6eb2e5b2c07 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -2366,13 +2366,15 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		dsp->running = false;
 
 		regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
-				   ADSP2_SYS_ENA | ADSP2_CORE_ENA |
-				   ADSP2_START, 0);
+				   ADSP2_CORE_ENA | ADSP2_START, 0);
 
 		/* Make sure DMAs are quiesced */
+		regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0);
 		regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0);
 		regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_2, 0);
-		regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0);
+
+		regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
+				   ADSP2_SYS_ENA, 0);
 
 		list_for_each_entry(ctl, &dsp->ctl_list, list)
 			ctl->enabled = 0;
-- 
cgit v1.2.3