From 22363e758770fa24edcc67ba33326f007bcd66a8 Mon Sep 17 00:00:00 2001
From: Xiubo Li <lixiubo@cmss.chinamobile.com>
Date: Tue, 19 Jan 2016 11:13:12 +0800
Subject: ASoC: ab8500: remove the useless 'break' after 'return'

'break' here is not useful after 'return' or 'goto'.

Signed-off-by: Xiubo Li <lixiubo@cmss.chinamobile.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index affb192..b6820a1 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2134,7 +2134,6 @@ static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 			"%s: ERROR: Unsupporter master mask 0x%x\n",
 			__func__, fmt & SND_SOC_DAIFMT_MASTER_MASK);
 		return -EINVAL;
-		break;
 	}
 
 	snd_soc_update_bits(codec, AB8500_DIGIFCONF3, mask, val);
-- 
cgit v0.10.2


From 0f9aa09d669b29ae536d51631bf035ad44ae0551 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 21 Jan 2016 17:52:57 +0000
Subject: ASoC: wm5110: Fix up snd_soc_register_platform error path

Whilst there is nothing wrong with the error path here it looks a little
bit odd, this patches makes the code a little more idiomatic.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 6088d30..83760b7 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2366,7 +2366,7 @@ static int wm5110_probe(struct platform_device *pdev)
 	ret = snd_soc_register_platform(&pdev->dev, &wm5110_compr_platform);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "Failed to register platform: %d\n", ret);
-		goto error;
+		return ret;
 	}
 
 	ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110,
@@ -2376,7 +2376,6 @@ static int wm5110_probe(struct platform_device *pdev)
 		snd_soc_unregister_platform(&pdev->dev);
 	}
 
-error:
 	return ret;
 }
 
-- 
cgit v0.10.2


From e6d00f3403c439b8d1e6f9f09d5ae3048aa726e4 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 21 Jan 2016 17:52:58 +0000
Subject: ASoC: wm_adsp: Use more generic naming for regions and caps
 definitions

The region definition will be shared by most firmwares so give this a
more generic name and whilst we are there improve the naming of the
voice control capabilities array as well.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 33806d4..0392c58 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -328,7 +328,7 @@ struct wm_adsp_buffer_region_def {
 	unsigned int size_offset;
 };
 
-static struct wm_adsp_buffer_region_def ez2control_regions[] = {
+static struct wm_adsp_buffer_region_def default_regions[] = {
 	{
 		.mem_type = WMFW_ADSP2_XM,
 		.base_offset = HOST_BUFFER_FIELD(X_buf_base),
@@ -353,7 +353,7 @@ struct wm_adsp_fw_caps {
 	struct wm_adsp_buffer_region_def *region_defs;
 };
 
-static const struct wm_adsp_fw_caps ez2control_caps[] = {
+static const struct wm_adsp_fw_caps ctrl_caps[] = {
 	{
 		.id = SND_AUDIOCODEC_BESPOKE,
 		.desc = {
@@ -362,8 +362,8 @@ static const struct wm_adsp_fw_caps ez2control_caps[] = {
 			.num_sample_rates = 1,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
 		},
-		.num_regions = ARRAY_SIZE(ez2control_regions),
-		.region_defs = ez2control_regions,
+		.num_regions = ARRAY_SIZE(default_regions),
+		.region_defs = default_regions,
 	},
 };
 
@@ -382,8 +382,8 @@ static const struct {
 	[WM_ADSP_FW_CTRL] =     {
 		.file = "ctrl",
 		.compr_direction = SND_COMPRESS_CAPTURE,
-		.num_caps = ARRAY_SIZE(ez2control_caps),
-		.caps = ez2control_caps,
+		.num_caps = ARRAY_SIZE(ctrl_caps),
+		.caps = ctrl_caps,
 	},
 	[WM_ADSP_FW_ASR] =      { .file = "asr" },
 	[WM_ADSP_FW_TRACE] =    { .file = "trace" },
-- 
cgit v0.10.2


From 7ce4283ca405d459c186960ed39d41ec0e6fb2bf Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 21 Jan 2016 17:52:59 +0000
Subject: ASoC: wm_adsp: Add debug audio trace firmware

The audio trace firmware allows the capture of arbitrary streams of
audio from the DSP and commonly used for debugging other firmwares. This
patch adds support for this firwmare into the ADSP driver.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 0392c58..a81f568 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -367,6 +367,24 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = {
 	},
 };
 
+static const struct wm_adsp_fw_caps trace_caps[] = {
+	{
+		.id = SND_AUDIOCODEC_BESPOKE,
+		.desc = {
+			.max_ch = 8,
+			.sample_rates = {
+				4000, 8000, 11025, 12000, 16000, 22050,
+				24000, 32000, 44100, 48000, 64000, 88200,
+				96000, 176400, 192000
+			},
+			.num_sample_rates = 15,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.num_regions = ARRAY_SIZE(default_regions),
+		.region_defs = default_regions,
+	},
+};
+
 static const struct {
 	const char *file;
 	int compr_direction;
@@ -386,7 +404,12 @@ static const struct {
 		.caps = ctrl_caps,
 	},
 	[WM_ADSP_FW_ASR] =      { .file = "asr" },
-	[WM_ADSP_FW_TRACE] =    { .file = "trace" },
+	[WM_ADSP_FW_TRACE] =    {
+		.file = "trace",
+		.compr_direction = SND_COMPRESS_CAPTURE,
+		.num_caps = ARRAY_SIZE(trace_caps),
+		.caps = trace_caps,
+	},
 	[WM_ADSP_FW_SPK_PROT] = { .file = "spk-prot" },
 	[WM_ADSP_FW_MISC] =     { .file = "misc" },
 };
-- 
cgit v0.10.2


From d82d767f0e61da5f1f872d40358904eb316af264 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 21 Jan 2016 17:53:02 +0000
Subject: ASoC: wm_adsp: Move setting of DSP speed into CODEC specific code

The ADSP code should be agnostic of which CODEC it runs upon, currently
there is only one remaining part of the implementation that doesn't
follow this. When the DSP is booted on ADSP2 we read
ARIZONA_SYSTEM_CLOCK_1 and use that to set the initial speed for the DSP
clock. This patch factors that out into CODEC specific code, leaving the
ADSP code entirely CODEC agnostic.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index dc5ae7f..9dc77a4 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -57,6 +57,25 @@ static const struct wm_adsp_region *cs47l24_dsp_regions[] = {
 	cs47l24_dsp3_regions,
 };
 
+static int cs47l24_adsp_power_ev(struct snd_soc_dapm_widget *w,
+				 struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+	unsigned int v;
+	int ret;
+
+	ret = regmap_read(arizona->regmap, ARIZONA_SYSTEM_CLOCK_1, &v);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to read SYSCLK state: %d\n", ret);
+		return ret;
+	}
+
+	v = (v & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT;
+
+	return wm_adsp2_early_event(w, kcontrol, event, v);
+}
+
 static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
 static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0);
@@ -405,8 +424,8 @@ SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
 SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
 		 NULL, 0),
 
-WM_ADSP2("DSP2", 1),
-WM_ADSP2("DSP3", 2),
+WM_ADSP2("DSP2", 1, cs47l24_adsp_power_ev),
+WM_ADSP2("DSP3", 2, cs47l24_adsp_power_ev),
 
 SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3,
 		 ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0),
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 64637d1..7c0860d 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -619,7 +619,7 @@ static int wm5102_adsp_power_ev(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
 	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
-	unsigned int v;
+	unsigned int v = 0;
 	int ret;
 
 	switch (event) {
@@ -654,7 +654,7 @@ static int wm5102_adsp_power_ev(struct snd_soc_dapm_widget *w,
 		break;
 	}
 
-	return wm_adsp2_early_event(w, kcontrol, event);
+	return wm_adsp2_early_event(w, kcontrol, event, v);
 }
 
 static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol,
@@ -1408,7 +1408,7 @@ ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
 ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
 ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
 
-WM_ADSP2_E("DSP1", 0, wm5102_adsp_power_ev),
+WM_ADSP2("DSP1", 0, wm5102_adsp_power_ev),
 
 SND_SOC_DAPM_OUTPUT("HPOUT1L"),
 SND_SOC_DAPM_OUTPUT("HPOUT1R"),
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 83760b7..b47bc44 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -191,6 +191,25 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int wm5110_adsp_power_ev(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+	unsigned int v;
+	int ret;
+
+	ret = regmap_read(arizona->regmap, ARIZONA_SYSTEM_CLOCK_1, &v);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to read SYSCLK state: %d\n", ret);
+		return ret;
+	}
+
+	v = (v & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT;
+
+	return wm_adsp2_early_event(w, kcontrol, event, v);
+}
+
 static const struct reg_sequence wm5110_no_dre_left_enable[] = {
 	{ 0x3024, 0xE410 },
 	{ 0x3025, 0x0056 },
@@ -1179,10 +1198,10 @@ SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
 SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
 		 NULL, 0),
 
-WM_ADSP2("DSP1", 0),
-WM_ADSP2("DSP2", 1),
-WM_ADSP2("DSP3", 2),
-WM_ADSP2("DSP4", 3),
+WM_ADSP2("DSP1", 0, wm5110_adsp_power_ev),
+WM_ADSP2("DSP2", 1, wm5110_adsp_power_ev),
+WM_ADSP2("DSP3", 2, wm5110_adsp_power_ev),
+WM_ADSP2("DSP4", 3, wm5110_adsp_power_ev),
 
 SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3,
 		 ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0),
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index a81f568..76ab52d 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -32,9 +32,6 @@
 #include <sound/initval.h>
 #include <sound/tlv.h>
 
-#include <linux/mfd/arizona/registers.h>
-
-#include "arizona.h"
 #include "wm_adsp.h"
 
 #define adsp_crit(_dsp, fmt, ...) \
@@ -2146,30 +2143,9 @@ static void wm_adsp2_boot_work(struct work_struct *work)
 					   struct wm_adsp,
 					   boot_work);
 	int ret;
-	unsigned int val;
 
 	mutex_lock(&dsp->pwr_lock);
 
-	/*
-	 * For simplicity set the DSP clock rate to be the
-	 * SYSCLK rate rather than making it configurable.
-	 */
-	ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val);
-	if (ret != 0) {
-		adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret);
-		goto err_mutex;
-	}
-	val = (val & ARIZONA_SYSCLK_FREQ_MASK)
-		>> ARIZONA_SYSCLK_FREQ_SHIFT;
-
-	ret = regmap_update_bits_async(dsp->regmap,
-				       dsp->base + ADSP2_CLOCKING,
-				       ADSP2_CLK_SEL_MASK, val);
-	if (ret != 0) {
-		adsp_err(dsp, "Failed to set clock rate: %d\n", ret);
-		goto err_mutex;
-	}
-
 	ret = wm_adsp2_ena(dsp);
 	if (ret != 0)
 		goto err_mutex;
@@ -2209,8 +2185,21 @@ err_mutex:
 	mutex_unlock(&dsp->pwr_lock);
 }
 
+static void wm_adsp2_set_dspclk(struct wm_adsp *dsp, unsigned int freq)
+{
+	int ret;
+
+	ret = regmap_update_bits_async(dsp->regmap,
+				       dsp->base + ADSP2_CLOCKING,
+				       ADSP2_CLK_SEL_MASK,
+				       freq << ADSP2_CLK_SEL_SHIFT);
+	if (ret != 0)
+		adsp_err(dsp, "Failed to set clock rate: %d\n", ret);
+}
+
 int wm_adsp2_early_event(struct snd_soc_dapm_widget *w,
-		   struct snd_kcontrol *kcontrol, int event)
+			 struct snd_kcontrol *kcontrol, int event,
+			 unsigned int freq)
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
 	struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
@@ -2220,6 +2209,7 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
+		wm_adsp2_set_dspclk(dsp, freq);
 		queue_work(system_unbound_wq, &dsp->boot_work);
 		break;
 	default:
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index 1a928ec..b61cb57 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -80,7 +80,7 @@ struct wm_adsp {
 	SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \
 		wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)
 
-#define WM_ADSP2_E(wname, num, event_fn) \
+#define WM_ADSP2(wname, num, event_fn) \
 {	.id = snd_soc_dapm_dai_link, .name = wname " Preloader", \
 	.reg = SND_SOC_NOPM, .shift = num, .event = event_fn, \
 	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }, \
@@ -88,9 +88,6 @@ struct wm_adsp {
 	.reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \
 	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD }
 
-#define WM_ADSP2(wname, num) \
-	WM_ADSP2_E(wname, num, wm_adsp2_early_event)
-
 extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
 
 int wm_adsp1_init(struct wm_adsp *dsp);
@@ -100,7 +97,8 @@ int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec);
 int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
 int wm_adsp2_early_event(struct snd_soc_dapm_widget *w,
-			 struct snd_kcontrol *kcontrol, int event);
+			 struct snd_kcontrol *kcontrol, int event,
+			 unsigned int freq);
 int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
 
-- 
cgit v0.10.2


From 31889507fd84a6a1c8dce36a214a070cec1fc559 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 21 Jan 2016 17:53:00 +0000
Subject: ASoC: wm5110: Add support for audio trace firmware

Mainly this adds DAI links for the audio trace, however, it is also
necessary to update the data IRQ handler to check more cores. We have
the handler check every core so it should not be necessary to update
this function if more compressed firmwares are added in the future.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 8b6adb5..0c64a57 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -57,7 +57,7 @@
 #define ARIZONA_CLK_98MHZ  5
 #define ARIZONA_CLK_147MHZ 6
 
-#define ARIZONA_MAX_DAI  8
+#define ARIZONA_MAX_DAI  10
 #define ARIZONA_MAX_ADSP 4
 
 #define ARIZONA_DVFS_SR1_RQ	0x001
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index b47bc44..92b6a6a 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1828,6 +1828,9 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "Voice Control DSP", NULL, "DSP3" },
 	{ "Voice Control DSP", NULL, "SYSCLK" },
 
+	{ "Audio Trace DSP", NULL, "DSP1" },
+	{ "Audio Trace DSP", NULL, "SYSCLK" },
+
 	{ "IN1L PGA", NULL, "IN1L" },
 	{ "IN1R PGA", NULL, "IN1R" },
 
@@ -2171,6 +2174,27 @@ static struct snd_soc_dai_driver wm5110_dai[] = {
 			.formats = WM5110_FORMATS,
 		},
 	},
+	{
+		.name = "wm5110-cpu-trace",
+		.capture = {
+			.stream_name = "Audio Trace CPU",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+		.compress_new = snd_soc_new_compress,
+	},
+	{
+		.name = "wm5110-dsp-trace",
+		.capture = {
+			.stream_name = "Audio Trace DSP",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+	},
 };
 
 static int wm5110_open(struct snd_compr_stream *stream)
@@ -2182,6 +2206,8 @@ static int wm5110_open(struct snd_compr_stream *stream)
 
 	if (strcmp(rtd->codec_dai->name, "wm5110-dsp-voicectrl") == 0) {
 		n_adsp = 2;
+	} else if (strcmp(rtd->codec_dai->name, "wm5110-dsp-trace") == 0) {
+		n_adsp = 0;
 	} else {
 		dev_err(arizona->dev,
 			"No suitable compressed stream for DAI '%s'\n",
@@ -2194,12 +2220,21 @@ static int wm5110_open(struct snd_compr_stream *stream)
 
 static irqreturn_t wm5110_adsp2_irq(int irq, void *data)
 {
-	struct wm5110_priv *florida = data;
-	int ret;
+	struct wm5110_priv *priv = data;
+	struct arizona *arizona = priv->core.arizona;
+	int serviced = 0;
+	int i, ret;
+
+	for (i = 0; i < WM5110_NUM_ADSP; ++i) {
+		ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
+		if (ret != -ENODEV)
+			serviced++;
+	}
 
-	ret = wm_adsp_compr_handle_irq(&florida->core.adsp[2]);
-	if (ret == -ENODEV)
+	if (!serviced) {
+		dev_err(arizona->dev, "Spurious compressed data IRQ\n");
 		return IRQ_NONE;
+	}
 
 	return IRQ_HANDLED;
 }
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 76ab52d..653dbff 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -2823,7 +2823,6 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
 	mutex_lock(&dsp->pwr_lock);
 
 	if (!buf) {
-		adsp_err(dsp, "Spurious buffer IRQ\n");
 		ret = -ENODEV;
 		goto out;
 	}
-- 
cgit v0.10.2


From 69fa80584f69554275ef1257d41b90f4e0d417fc Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 21 Jan 2016 17:53:01 +0000
Subject: ASoC: wm5102: Add support for the audio trace firmware

wm5102 also supports the audio trace firmware, this patch adds support
for this into the wm5102 driver.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 7c0860d..aa1e96a 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1599,6 +1599,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	{ "Slim2 Capture", NULL, "SYSCLK" },
 	{ "Slim3 Capture", NULL, "SYSCLK" },
 
+	{ "Audio Trace DSP", NULL, "DSP1" },
+	{ "Audio Trace DSP", NULL, "SYSCLK" },
+
 	{ "IN1L PGA", NULL, "IN1L" },
 	{ "IN1R PGA", NULL, "IN1R" },
 
@@ -1864,14 +1867,67 @@ static struct snd_soc_dai_driver wm5102_dai[] = {
 		 },
 		.ops = &arizona_simple_dai_ops,
 	},
+	{
+		.name = "wm5102-cpu-trace",
+		.capture = {
+			.stream_name = "Audio Trace CPU",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+		.compress_new = snd_soc_new_compress,
+	},
+	{
+		.name = "wm5102-dsp-trace",
+		.capture = {
+			.stream_name = "Audio Trace DSP",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+	},
 };
 
+static int wm5102_open(struct snd_compr_stream *stream)
+{
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	struct wm5102_priv *priv = snd_soc_codec_get_drvdata(rtd->codec);
+
+	return wm_adsp_compr_open(&priv->core.adsp[0], stream);
+}
+
+static irqreturn_t wm5102_adsp2_irq(int irq, void *data)
+{
+	struct wm5102_priv *priv = data;
+	struct arizona *arizona = priv->core.arizona;
+	int ret;
+
+	ret = wm_adsp_compr_handle_irq(&priv->core.adsp[0]);
+	if (ret == -ENODEV) {
+		dev_err(arizona->dev, "Spurious compressed data IRQ\n");
+		return IRQ_NONE;
+	}
+
+	return IRQ_HANDLED;
+}
+
 static int wm5102_codec_probe(struct snd_soc_codec *codec)
 {
 	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->core.arizona;
 	int ret;
 
+	ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1,
+				  "ADSP2 Compressed IRQ", wm5102_adsp2_irq,
+				  priv);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request DSP IRQ: %d\n", ret);
+		return ret;
+	}
+
 	ret = wm_adsp2_codec_probe(&priv->core.adsp[0], codec);
 	if (ret)
 		return ret;
@@ -1946,6 +2002,20 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5102 = {
 	.num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes),
 };
 
+static struct snd_compr_ops wm5102_compr_ops = {
+	.open = wm5102_open,
+	.free = wm_adsp_compr_free,
+	.set_params = wm_adsp_compr_set_params,
+	.get_caps = wm_adsp_compr_get_caps,
+	.trigger = wm_adsp_compr_trigger,
+	.pointer = wm_adsp_compr_pointer,
+	.copy = wm_adsp_compr_copy,
+};
+
+static struct snd_soc_platform_driver wm5102_compr_platform = {
+	.compr_ops = &wm5102_compr_ops,
+};
+
 static int wm5102_probe(struct platform_device *pdev)
 {
 	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
@@ -2005,12 +2075,25 @@ static int wm5102_probe(struct platform_device *pdev)
 	pm_runtime_enable(&pdev->dev);
 	pm_runtime_idle(&pdev->dev);
 
-	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102,
+	ret = snd_soc_register_platform(&pdev->dev, &wm5102_compr_platform);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register platform: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102,
 				      wm5102_dai, ARRAY_SIZE(wm5102_dai));
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
+		snd_soc_unregister_platform(&pdev->dev);
+	}
+
+	return ret;
 }
 
 static int wm5102_remove(struct platform_device *pdev)
 {
+	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
-- 
cgit v0.10.2


From 7eec55d7a2f2932b3b4a799bc5d0d6718e1ee9e2 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Date: Fri, 22 Jan 2016 13:49:59 +0000
Subject: ASoC: cs47l24: Add voice control compressed stream

This patch adds firmware compressed stream capture support
and DAI hookups for voice control firmware (based on the wm5110
implementation.)

Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 9dc77a4..b8d3cd0 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -798,6 +798,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = {
 	{ "AIF2 Capture", NULL, "SYSCLK" },
 	{ "AIF3 Capture", NULL, "SYSCLK" },
 
+	{ "Voice Control DSP", NULL, "DSP3" },
+	{ "Voice Control DSP", NULL, "SYSCLK" },
+
 	{ "IN1L PGA", NULL, "IN1L" },
 	{ "IN1R PGA", NULL, "IN1R" },
 
@@ -992,12 +995,68 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
 		.symmetric_rates = 1,
 		.symmetric_samplebits = 1,
 	},
+	{
+		.name = "cs47l24-cpu-voicectrl",
+		.capture = {
+			.stream_name = "Voice Control CPU",
+			.channels_min = 1,
+			.channels_max = 1,
+			.rates = CS47L24_RATES,
+			.formats = CS47L24_FORMATS,
+		},
+		.compress_new = snd_soc_new_compress,
+	},
+	{
+		.name = "cs47l24-dsp-voicectrl",
+		.capture = {
+			.stream_name = "Voice Control DSP",
+			.channels_min = 1,
+			.channels_max = 1,
+			.rates = CS47L24_RATES,
+			.formats = CS47L24_FORMATS,
+		},
+	},
 };
 
+static int cs47l24_open(struct snd_compr_stream *stream)
+{
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(rtd->codec);
+	struct arizona *arizona = priv->core.arizona;
+	int n_adsp;
+
+	if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) {
+		n_adsp = 2;
+	} else {
+		dev_err(arizona->dev,
+			"No suitable compressed stream for DAI '%s'\n",
+			rtd->codec_dai->name);
+		return -EINVAL;
+	}
+
+	return wm_adsp_compr_open(&priv->core.adsp[n_adsp], stream);
+}
+
+static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
+{
+	struct cs47l24_priv *priv = data;
+	struct arizona *arizona = priv->core.arizona;
+	int ret;
+
+	ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]);
+	if (ret == -ENODEV) {
+		dev_err(arizona->dev, "Spurious compressed data IRQ\n");
+		return IRQ_NONE;
+	}
+
+	return IRQ_HANDLED;
+}
+
 static int cs47l24_codec_probe(struct snd_soc_codec *codec)
 {
 	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->core.arizona;
 	int ret;
 
 	priv->core.arizona->dapm = dapm;
@@ -1006,6 +1065,14 @@ static int cs47l24_codec_probe(struct snd_soc_codec *codec)
 	arizona_init_gpio(codec);
 	arizona_init_mono(codec);
 
+	ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1,
+				  "ADSP2 Compressed IRQ", cs47l24_adsp2_irq,
+				  priv);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request DSP IRQ: %d\n", ret);
+		return ret;
+	}
+
 	ret = wm_adsp2_codec_probe(&priv->core.adsp[1], codec);
 	if (ret)
 		goto err_adsp2_codec_probe;
@@ -1033,13 +1100,14 @@ err_adsp2_codec_probe:
 static int cs47l24_codec_remove(struct snd_soc_codec *codec)
 {
 	struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(codec);
-
+	struct arizona *arizona = priv->core.arizona;
 
 	wm_adsp2_codec_remove(&priv->core.adsp[1], codec);
 	wm_adsp2_codec_remove(&priv->core.adsp[2], codec);
 
 	priv->core.arizona->dapm = NULL;
 
+	arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv);
 	return 0;
 }
 
@@ -1076,6 +1144,19 @@ static struct snd_soc_codec_driver soc_codec_dev_cs47l24 = {
 	.num_dapm_routes = ARRAY_SIZE(cs47l24_dapm_routes),
 };
 
+static struct snd_compr_ops cs47l24_compr_ops = {
+	.open = cs47l24_open,
+	.free = wm_adsp_compr_free,
+	.set_params = wm_adsp_compr_set_params,
+	.get_caps = wm_adsp_compr_get_caps,
+	.trigger = wm_adsp_compr_trigger,
+	.pointer = wm_adsp_compr_pointer,
+	.copy = wm_adsp_compr_copy,
+};
+
+static struct snd_soc_platform_driver cs47l24_compr_platform = {
+	.compr_ops = &cs47l24_compr_ops,
+};
 static int cs47l24_probe(struct platform_device *pdev)
 {
 	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
@@ -1139,12 +1220,25 @@ static int cs47l24_probe(struct platform_device *pdev)
 	pm_runtime_enable(&pdev->dev);
 	pm_runtime_idle(&pdev->dev);
 
-	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24,
+	ret = snd_soc_register_platform(&pdev->dev, &cs47l24_compr_platform);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register platform: %d\n", ret);
+		return ret;
+	}
+	ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24,
 				      cs47l24_dai, ARRAY_SIZE(cs47l24_dai));
+
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
+		snd_soc_unregister_platform(&pdev->dev);
+	}
+
+	return ret;
 }
 
 static int cs47l24_remove(struct platform_device *pdev)
 {
+	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
-- 
cgit v0.10.2


From 3a9686c4ba7eec9201c6e8793705d32f24748771 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Mon, 1 Feb 2016 15:22:34 +0000
Subject: ASoC: wm_adsp: Add missing const from region definitions

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 653dbff..28d3851 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -325,7 +325,7 @@ struct wm_adsp_buffer_region_def {
 	unsigned int size_offset;
 };
 
-static struct wm_adsp_buffer_region_def default_regions[] = {
+static const struct wm_adsp_buffer_region_def default_regions[] = {
 	{
 		.mem_type = WMFW_ADSP2_XM,
 		.base_offset = HOST_BUFFER_FIELD(X_buf_base),
@@ -347,7 +347,7 @@ struct wm_adsp_fw_caps {
 	u32 id;
 	struct snd_codec_desc desc;
 	int num_regions;
-	struct wm_adsp_buffer_region_def *region_defs;
+	const struct wm_adsp_buffer_region_def *region_defs;
 };
 
 static const struct wm_adsp_fw_caps ctrl_caps[] = {
-- 
cgit v0.10.2


From da2b33589013b4cb3dad163f50fae060896cb8b6 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Tue, 2 Feb 2016 16:41:36 +0000
Subject: ASoC: wm_adsp: Return sample rate in wm_adsp_compr_pointer

We should return a valid sample rate from the pointer callback, this
patch adds this into the driver.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 33806d4..68b85ee 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -295,6 +295,8 @@ struct wm_adsp_compr {
 
 	u32 *raw_buf;
 	unsigned int copied_total;
+
+	unsigned int sample_rate;
 };
 
 #define WM_ADSP_DATA_WORD_SIZE         3
@@ -2471,6 +2473,8 @@ int wm_adsp_compr_set_params(struct snd_compr_stream *stream,
 	if (!compr->raw_buf)
 		return -ENOMEM;
 
+	compr->sample_rate = params->codec.sample_rate;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp_compr_set_params);
@@ -2911,6 +2915,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
 
 	tstamp->copied_total = compr->copied_total;
 	tstamp->copied_total += buf->avail * WM_ADSP_DATA_WORD_SIZE;
+	tstamp->sampling_rate = compr->sample_rate;
 
 out:
 	mutex_unlock(&dsp->pwr_lock);
-- 
cgit v0.10.2


From d81221ff9406a3918084bfaed752894e82c5d44b Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 4 Feb 2016 16:29:01 +0000
Subject: ASoC: arizona: Add support for SNDRV_PCM_RATE_KNOT

The Arizona CODECs support several rates that do not have simple defines
in ALSA. This patch adds support for SNDRV_PCM_RATE_KNOT so that users
can open stream at these rates. As part of this we should always set
constraints in arizona_startup, currently we only set the constraints if
we already have a clock to limit rates to that family of sample rates.
This patch updates this to set a constraint of all rates supported by
the chip if we do not already know which family of rates to limit to.
Finally we also reduce the list of rates supported in the constraints to
only include those that are supported on current parts.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 33143fe..a32cfb8 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1398,29 +1398,6 @@ static const int arizona_48k_bclk_rates[] = {
 	24576000,
 };
 
-static const unsigned int arizona_48k_rates[] = {
-	12000,
-	24000,
-	48000,
-	96000,
-	192000,
-	384000,
-	768000,
-	4000,
-	8000,
-	16000,
-	32000,
-	64000,
-	128000,
-	256000,
-	512000,
-};
-
-static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = {
-	.count	= ARRAY_SIZE(arizona_48k_rates),
-	.list	= arizona_48k_rates,
-};
-
 static const int arizona_44k1_bclk_rates[] = {
 	-1,
 	44100,
@@ -1443,22 +1420,7 @@ static const int arizona_44k1_bclk_rates[] = {
 	22579200,
 };
 
-static const unsigned int arizona_44k1_rates[] = {
-	11025,
-	22050,
-	44100,
-	88200,
-	176400,
-	352800,
-	705600,
-};
-
-static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = {
-	.count	= ARRAY_SIZE(arizona_44k1_rates),
-	.list	= arizona_44k1_rates,
-};
-
-static int arizona_sr_vals[] = {
+static const unsigned int arizona_sr_vals[] = {
 	0,
 	12000,
 	24000,
@@ -1485,13 +1447,21 @@ static int arizona_sr_vals[] = {
 	512000,
 };
 
+#define ARIZONA_48K_RATE_MASK	0x0F003E
+#define ARIZONA_44K1_RATE_MASK	0x003E00
+#define ARIZONA_RATE_MASK	(ARIZONA_48K_RATE_MASK | ARIZONA_44K1_RATE_MASK)
+
+static const struct snd_pcm_hw_constraint_list arizona_constraint = {
+	.count	= ARRAY_SIZE(arizona_sr_vals),
+	.list	= arizona_sr_vals,
+};
+
 static int arizona_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
-	const struct snd_pcm_hw_constraint_list *constraint;
 	unsigned int base_rate;
 
 	if (!substream->runtime)
@@ -1509,16 +1479,15 @@ static int arizona_startup(struct snd_pcm_substream *substream,
 	}
 
 	if (base_rate == 0)
-		return 0;
-
-	if (base_rate % 8000)
-		constraint = &arizona_44k1_constraint;
+		dai_priv->constraint.mask = ARIZONA_RATE_MASK;
+	else if (base_rate % 8000)
+		dai_priv->constraint.mask = ARIZONA_44K1_RATE_MASK;
 	else
-		constraint = &arizona_48k_constraint;
+		dai_priv->constraint.mask = ARIZONA_48K_RATE_MASK;
 
 	return snd_pcm_hw_constraint_list(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_RATE,
-					  constraint);
+					  &dai_priv->constraint);
 }
 
 static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec,
@@ -1911,6 +1880,7 @@ int arizona_init_dai(struct arizona_priv *priv, int id)
 	struct arizona_dai_priv *dai_priv = &priv->dai[id];
 
 	dai_priv->clk = ARIZONA_CLK_SYSCLK;
+	dai_priv->constraint = arizona_constraint;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 0c64a57..1ea8e4e 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -68,6 +68,8 @@ struct wm_adsp;
 
 struct arizona_dai_priv {
 	int clk;
+
+	struct snd_pcm_hw_constraint_list constraint;
 };
 
 struct arizona_priv {
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index b8d3cd0..576087b 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -923,7 +923,7 @@ static int cs47l24_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	}
 }
 
-#define CS47L24_RATES SNDRV_PCM_RATE_8000_192000
+#define CS47L24_RATES SNDRV_PCM_RATE_KNOT
 
 #define CS47L24_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index aa1e96a..a8b3e3f 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1738,7 +1738,7 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	}
 }
 
-#define WM5102_RATES SNDRV_PCM_RATE_8000_192000
+#define WM5102_RATES SNDRV_PCM_RATE_KNOT
 
 #define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 92b6a6a..aa47955 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2024,7 +2024,7 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	}
 }
 
-#define WM5110_RATES SNDRV_PCM_RATE_8000_192000
+#define WM5110_RATES SNDRV_PCM_RATE_KNOT
 
 #define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index b4dba3a..52d766e 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -943,7 +943,7 @@ static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	}
 }
 
-#define WM8997_RATES SNDRV_PCM_RATE_8000_192000
+#define WM8997_RATES SNDRV_PCM_RATE_KNOT
 
 #define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 7719bc5..0123960 100644
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -1170,7 +1170,7 @@ static const struct snd_soc_dapm_route wm8998_dapm_routes[] = {
 	{ "DRC1 Signal Activity", NULL, "DRC1R" },
 };
 
-#define WM8998_RATES SNDRV_PCM_RATE_8000_192000
+#define WM8998_RATES SNDRV_PCM_RATE_KNOT
 
 #define WM8998_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-- 
cgit v0.10.2


From 27d6e7d1c96c9f51379e0feb972fec26029098bc Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Andreas=20Irest=C3=A5l?= <andreas.irestal@axis.com>
Date: Thu, 4 Feb 2016 15:05:19 +0100
Subject: ASoC: adau17x1: Cache writes when core clock is disabled
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

In some configurations, the dai registers get written before the bias
level is changed in the codec driver. This leads to a situation where
an initial write to the serial port register gets ignored, and future
writes may as well, since regmap thinks that the codec already holds the
value. More specifically, configuring the codec as i2s master would in
fact result in the codec running as slave, a situation where no i2s
clocks are generated and hence no data is transferred.

This change makes sure that regmap only caches writes when the core
clock is disabled, and syncs regmap whenever enabling the core clock
again.

Signed-off-by: Andreas Irestål <andire@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index 2f12477..e7136b1 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -456,13 +456,17 @@ static int adau1761_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
+		regcache_cache_only(adau->regmap, false);
 		regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL,
 			ADAU17X1_CLOCK_CONTROL_SYSCLK_EN,
 			ADAU17X1_CLOCK_CONTROL_SYSCLK_EN);
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
+			regcache_sync(adau->regmap);
 		break;
 	case SND_SOC_BIAS_OFF:
 		regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL,
 			ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, 0);
+		regcache_cache_only(adau->regmap, true);
 		break;
 
 	}
@@ -783,6 +787,10 @@ int adau1761_probe(struct device *dev, struct regmap *regmap,
 	if (ret)
 		return ret;
 
+	/* Enable cache only mode as we could miss writes before bias level
+	 * reaches standby and the core clock is enabled */
+	regcache_cache_only(regmap, true);
+
 	return snd_soc_register_codec(dev, &adau1761_codec_driver, dai_drv, 1);
 }
 EXPORT_SYMBOL_GPL(adau1761_probe);
-- 
cgit v0.10.2


From c7dae7c4c60693286298e300b3d140c8cff46daa Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 19 Feb 2016 14:44:41 +0000
Subject: ASoC: wm_adsp: Avoid narrow race condition on compr pointer

There is a very small window between then wm_adsp_compr_free gets call
and when the DSP is actually powered down. If we get an IRQ from the DSP
in this window then the wm_adsp_compr pointer will be NULL. This patch
adds a check for this into the IRQ handler to avoid any issues when this
happens.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 68b85ee..cb49a63 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -2845,7 +2845,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
 		goto out;
 	}
 
-	if (compr->stream)
+	if (compr && compr->stream)
 		snd_compr_fragment_elapsed(compr->stream);
 
 out:
-- 
cgit v0.10.2


From cc815c4b86d14630a7d4ca4f4c7d2c2acb88cb52 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 19 Feb 2016 14:44:42 +0000
Subject: ASoC: wm_adsp: Explicitly set the control access flags in all cases

In the case where the firmware does not tell us the access flags for the
control, we let ALSA select a default (READWRITE). But really we should
be applying the volatile flag in this case, as we will read the control
from the DSP if it is on in this case. This patch explicitly sets the
access flags in all cases.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index cb49a63..ff0ce6ba 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -857,6 +857,9 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
 			kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ;
 		if (ctl->flags & WMFW_CTL_FLAG_VOLATILE)
 			kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+	} else {
+		kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+		kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
 	}
 
 	ret = snd_soc_add_card_controls(dsp->card,
-- 
cgit v0.10.2


From 7d00cd975a52e637da19548c1cc7d87a0a32b49c Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 19 Feb 2016 14:44:43 +0000
Subject: ASoC: wm_adsp: Fixup some odd line wrapping

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ff0ce6ba..0f6eeb1 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -862,15 +862,13 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
 		kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
 	}
 
-	ret = snd_soc_add_card_controls(dsp->card,
-					kcontrol, 1);
+	ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1);
 	if (ret < 0)
 		goto err_kcontrol;
 
 	kfree(kcontrol);
 
-	ctl->kcontrol = snd_soc_card_get_kcontrol(dsp->card,
-						  ctl->name);
+	ctl->kcontrol = snd_soc_card_get_kcontrol(dsp->card, ctl->name);
 
 	return 0;
 
@@ -890,9 +888,7 @@ static int wm_coeff_init_control_caches(struct wm_adsp *dsp)
 		if (ctl->flags & WMFW_CTL_FLAG_VOLATILE)
 			continue;
 
-		ret = wm_coeff_read_control(ctl,
-					    ctl->cache,
-					    ctl->len);
+		ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len);
 		if (ret < 0)
 			return ret;
 	}
@@ -909,9 +905,7 @@ static int wm_coeff_sync_controls(struct wm_adsp *dsp)
 		if (!ctl->enabled)
 			continue;
 		if (ctl->set && !(ctl->flags & WMFW_CTL_FLAG_VOLATILE)) {
-			ret = wm_coeff_write_control(ctl,
-						     ctl->cache,
-						     ctl->len);
+			ret = wm_coeff_write_control(ctl, ctl->cache, ctl->len);
 			if (ret < 0)
 				return ret;
 		}
@@ -1507,8 +1501,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs,
 
 	ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2);
 	if (ret != 0) {
-		adsp_err(dsp, "Failed to read algorithm list: %d\n",
-			ret);
+		adsp_err(dsp, "Failed to read algorithm list: %d\n", ret);
 		kfree(alg);
 		return ERR_PTR(ret);
 	}
@@ -2007,8 +2000,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 				goto err_mutex;
 			}
 
-			val = (val & dsp->sysclk_mask)
-				>> dsp->sysclk_shift;
+			val = (val & dsp->sysclk_mask) >> dsp->sysclk_shift;
 
 			ret = regmap_update_bits(dsp->regmap,
 						 dsp->base + ADSP1_CONTROL_31,
@@ -2101,8 +2093,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 
 	/* Wait for the RAM to start, should be near instantaneous */
 	for (count = 0; count < 10; ++count) {
-		ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
-				  &val);
+		ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1, &val);
 		if (ret != 0)
 			return ret;
 
-- 
cgit v0.10.2


From 4f8ea6d759f6ed5d448472143cef7681e319c958 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 19 Feb 2016 14:44:44 +0000
Subject: ASoC: wm_adsp: wm_coeff_{read|write}_control should use passed length

wm_coeff_{read|write}_control were using the control length rather than
the length parameter passed to them. This is not causing any issues as
the two values are currently always the same, but this needs fixed to
allow future improvements.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 0f6eeb1..36687db 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -721,19 +721,19 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl,
 	reg = ctl->alg_region.base + ctl->offset;
 	reg = wm_adsp_region_to_reg(mem, reg);
 
-	scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA);
+	scratch = kmemdup(buf, len, GFP_KERNEL | GFP_DMA);
 	if (!scratch)
 		return -ENOMEM;
 
 	ret = regmap_raw_write(dsp->regmap, reg, scratch,
-			       ctl->len);
+			       len);
 	if (ret) {
 		adsp_err(dsp, "Failed to write %zu bytes to %x: %d\n",
-			 ctl->len, reg, ret);
+			 len, reg, ret);
 		kfree(scratch);
 		return ret;
 	}
-	adsp_dbg(dsp, "Wrote %zu bytes to %x\n", ctl->len, reg);
+	adsp_dbg(dsp, "Wrote %zu bytes to %x\n", len, reg);
 
 	kfree(scratch);
 
@@ -780,20 +780,20 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
 	reg = ctl->alg_region.base + ctl->offset;
 	reg = wm_adsp_region_to_reg(mem, reg);
 
-	scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA);
+	scratch = kmalloc(len, GFP_KERNEL | GFP_DMA);
 	if (!scratch)
 		return -ENOMEM;
 
-	ret = regmap_raw_read(dsp->regmap, reg, scratch, ctl->len);
+	ret = regmap_raw_read(dsp->regmap, reg, scratch, len);
 	if (ret) {
 		adsp_err(dsp, "Failed to read %zu bytes from %x: %d\n",
 			 ctl->len, reg, ret);
 		kfree(scratch);
 		return ret;
 	}
-	adsp_dbg(dsp, "Read %zu bytes from %x\n", ctl->len, reg);
+	adsp_dbg(dsp, "Read %zu bytes from %x\n", len, reg);
 
-	memcpy(buf, scratch, ctl->len);
+	memcpy(buf, scratch, len);
 	kfree(scratch);
 
 	return 0;
-- 
cgit v0.10.2


From 2595b7fe1aa3a1dfc47e549b39f7c35c5ce8a56d Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Date: Tue, 23 Feb 2016 14:16:32 +0000
Subject: ASoC: arizona: Make logging of FLL calculations clearer

The debug logging of FLL calculations was confusing. Values were
printed in hex without indicating this by a leading 0x, and
despite these normally being required in decimal. Also where the
register value isn't the actual value (it s a power-of-two or
0 means 1, 1 means 2, ...)  it was unclear whether the actual or
register value was shown.

This patch changes the log print so that all mathematical values
are shown in decimal, all register values are shown in hex with a
0x prefix, and where the register value isn't the actual integer
value the decimal integer value is shown in () after the hex
register value.

Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index a32cfb8..dd926f4 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -2108,11 +2108,12 @@ static int arizona_calc_fll(struct arizona_fll *fll,
 		return -EINVAL;
 	}
 
-	arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n",
+	arizona_fll_dbg(fll, "N=%d THETA=%d LAMBDA=%d\n",
 			cfg->n, cfg->theta, cfg->lambda);
-	arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
-			cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv);
-	arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain);
+	arizona_fll_dbg(fll, "FRATIO=0x%x(%d) OUTDIV=%d REFCLK_DIV=0x%x(%d)\n",
+			cfg->fratio, ratio, cfg->outdiv,
+			cfg->refdiv, 1 << cfg->refdiv);
+	arizona_fll_dbg(fll, "GAIN=0x%x(%d)\n", cfg->gain, 1 << cfg->gain);
 
 	return 0;
 
-- 
cgit v0.10.2


From d6dde63e90cf0d34ae61e885a5eb254c59eb3358 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Andreas=20Irest=C3=A5l?= <andreas.irestal@axis.com>
Date: Tue, 16 Feb 2016 13:56:42 +0100
Subject: ASoC: adau17x1: Correct typos in file headers
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Signed-off-by: Andreas Irestål <andire@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/include/linux/platform_data/adau17x1.h b/include/linux/platform_data/adau17x1.h
index a81766c..9db1b90 100644
--- a/include/linux/platform_data/adau17x1.h
+++ b/include/linux/platform_data/adau17x1.h
@@ -1,5 +1,5 @@
 /*
- * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961/ADAU1781/ADAU1781 codecs
+ * Driver for ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781 codecs
  *
  * Copyright 2011-2014 Analog Devices Inc.
  * Author: Lars-Peter Clausen <lars@metafoo.de>
diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c
index 348ccb1..a1b1267 100644
--- a/sound/soc/codecs/adau1761-i2c.c
+++ b/sound/soc/codecs/adau1761-i2c.c
@@ -1,5 +1,5 @@
 /*
- * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961 codec
+ * Driver for ADAU1361/ADAU1461/ADAU1761/ADAU1961 codec
  *
  * Copyright 2014 Analog Devices Inc.
  *  Author: Lars-Peter Clausen <lars@metafoo.de>
diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c
index 8bc1fbd..1883349 100644
--- a/sound/soc/codecs/adau1761-spi.c
+++ b/sound/soc/codecs/adau1761-spi.c
@@ -1,5 +1,5 @@
 /*
- * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961 codec
+ * Driver for ADAU1361/ADAU1461/ADAU1761/ADAU1961 codec
  *
  * Copyright 2014 Analog Devices Inc.
  *  Author: Lars-Peter Clausen <lars@metafoo.de>
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index e7136b1..b95d29d 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -1,5 +1,5 @@
 /*
- * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961 codec
+ * Driver for ADAU1361/ADAU1461/ADAU1761/ADAU1961 codec
  *
  * Copyright 2011-2013 Analog Devices Inc.
  * Author: Lars-Peter Clausen <lars@metafoo.de>
diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c
index fde9068..bc1bb56 100644
--- a/sound/soc/codecs/adau1781.c
+++ b/sound/soc/codecs/adau1781.c
@@ -1,5 +1,5 @@
 /*
- * Driver for ADAU1781/ADAU1781 codec
+ * Driver for ADAU1381/ADAU1781 codec
  *
  * Copyright 2011-2013 Analog Devices Inc.
  * Author: Lars-Peter Clausen <lars@metafoo.de>
-- 
cgit v0.10.2


From ef329457237d885baf3408a56db9a02df97cde87 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Andreas=20Irest=C3=A5l?= <andreas.irestal@axis.com>
Date: Tue, 16 Feb 2016 13:56:43 +0100
Subject: ASoC: adau17x1: Add adau17x1 bindings document
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Signed-off-by: Andreas Irestål <andire@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
new file mode 100644
index 0000000..8dbce0e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt
@@ -0,0 +1,24 @@
+Analog Devices ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781
+
+Required properties:
+
+ - compatible:		Should contain one of the following:
+			"adi,adau1361"
+			"adi,adau1461"
+			"adi,adau1761"
+			"adi,adau1961"
+			"adi,adau1381"
+			"adi,adau1781"
+
+ - reg:			The i2c address. Value depends on the state of ADDR0
+			and ADDR1, as wired in hardware.
+
+Examples:
+#include <dt-bindings/sound/adau17x1.h>
+
+	i2c_bus {
+		adau1361@38 {
+			compatible = "adi,adau1761";
+			reg = <0x38>;
+		};
+	};
-- 
cgit v0.10.2


From aaf0f3aeeb6f9e5e874fccc8a0e56b00fea537c3 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Andreas=20Irest=C3=A5l?= <andreas.irestal@axis.com>
Date: Tue, 16 Feb 2016 13:56:44 +0100
Subject: ASoC: adau17x1: Add basic DT support for adau17x1
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Signed-off-by: Andreas Irestål <andire@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c
index a1b1267..8de010f 100644
--- a/sound/soc/codecs/adau1761-i2c.c
+++ b/sound/soc/codecs/adau1761-i2c.c
@@ -44,9 +44,21 @@ static const struct i2c_device_id adau1761_i2c_ids[] = {
 };
 MODULE_DEVICE_TABLE(i2c, adau1761_i2c_ids);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id adau1761_i2c_dt_ids[] = {
+	{ .compatible = "adi,adau1361", },
+	{ .compatible = "adi,adau1461", },
+	{ .compatible = "adi,adau1761", },
+	{ .compatible = "adi,adau1961", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, adau1761_i2c_dt_ids);
+#endif
+
 static struct i2c_driver adau1761_i2c_driver = {
 	.driver = {
 		.name = "adau1761",
+		.of_match_table = of_match_ptr(adau1761_i2c_dt_ids),
 	},
 	.probe = adau1761_i2c_probe,
 	.remove = adau1761_i2c_remove,
diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c
index 1883349..d917124 100644
--- a/sound/soc/codecs/adau1761-spi.c
+++ b/sound/soc/codecs/adau1761-spi.c
@@ -61,9 +61,21 @@ static const struct spi_device_id adau1761_spi_id[] = {
 };
 MODULE_DEVICE_TABLE(spi, adau1761_spi_id);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id adau1761_spi_dt_ids[] = {
+	{ .compatible = "adi,adau1361", },
+	{ .compatible = "adi,adau1461", },
+	{ .compatible = "adi,adau1761", },
+	{ .compatible = "adi,adau1961", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, adau1761_spi_dt_ids);
+#endif
+
 static struct spi_driver adau1761_spi_driver = {
 	.driver = {
 		.name = "adau1761",
+		.of_match_table = of_match_ptr(adau1761_spi_dt_ids),
 	},
 	.probe = adau1761_spi_probe,
 	.remove = adau1761_spi_remove,
diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c
index 0e32bba..06cbca8 100644
--- a/sound/soc/codecs/adau1781-i2c.c
+++ b/sound/soc/codecs/adau1781-i2c.c
@@ -42,9 +42,19 @@ static const struct i2c_device_id adau1781_i2c_ids[] = {
 };
 MODULE_DEVICE_TABLE(i2c, adau1781_i2c_ids);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id adau1781_i2c_dt_ids[] = {
+	{ .compatible = "adi,adau1381", },
+	{ .compatible = "adi,adau1781", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, adau1781_i2c_dt_ids);
+#endif
+
 static struct i2c_driver adau1781_i2c_driver = {
 	.driver = {
 		.name = "adau1781",
+		.of_match_table = of_match_ptr(adau1781_i2c_dt_ids),
 	},
 	.probe = adau1781_i2c_probe,
 	.remove = adau1781_i2c_remove,
diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c
index 33a73ff..3d965a0 100644
--- a/sound/soc/codecs/adau1781-spi.c
+++ b/sound/soc/codecs/adau1781-spi.c
@@ -59,9 +59,19 @@ static const struct spi_device_id adau1781_spi_id[] = {
 };
 MODULE_DEVICE_TABLE(spi, adau1781_spi_id);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id adau1781_spi_dt_ids[] = {
+	{ .compatible = "adi,adau1381", },
+	{ .compatible = "adi,adau1781", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, adau1781_spi_dt_ids);
+#endif
+
 static struct spi_driver adau1781_spi_driver = {
 	.driver = {
 		.name = "adau1781",
+		.of_match_table = of_match_ptr(adau1781_spi_dt_ids),
 	},
 	.probe = adau1781_spi_probe,
 	.remove = adau1781_spi_remove,
-- 
cgit v0.10.2


From c9146fae4e88fa9864c4a7087289523b96cdbf2d Mon Sep 17 00:00:00 2001
From: Florian Vaussard <florian.vaussard@gmail.com>
Date: Fri, 5 Feb 2016 16:32:13 +0100
Subject: ASoC: ads117x: Add bindings documentation for TI ADS117x ADC

Currently the binding is only made of the compatible string.

Signed-off-by: Florian Vaussard <florian.vaussard@heig-vd.ch>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/Documentation/devicetree/bindings/sound/ti,ads117x.txt b/Documentation/devicetree/bindings/sound/ti,ads117x.txt
new file mode 100644
index 0000000..7db19b5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ti,ads117x.txt
@@ -0,0 +1,11 @@
+Texas Intstruments ADS117x ADC
+
+Required properties:
+
+  - compatible : "ti,ads1174" or "ti,ads1178"
+
+Example:
+
+ads1178 {
+	compatible = "ti,ads1178";
+};
-- 
cgit v0.10.2


From 4f2bf0ace0f96cc693002e1bbde967fa2356bc43 Mon Sep 17 00:00:00 2001
From: Florian Vaussard <florian.vaussard@gmail.com>
Date: Fri, 5 Feb 2016 16:32:14 +0100
Subject: ASoC: ads117x: Add device tree compatible string

This patch adds the necessary device tree compatible string to allow DT
probing.

Signed-off-by: Florian Vaussard <florian.vaussard@heig-vd.ch>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c
index 1222282..c5be1bd 100644
--- a/sound/soc/codecs/ads117x.c
+++ b/sound/soc/codecs/ads117x.c
@@ -20,6 +20,8 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
+#include <linux/of.h>
+
 #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000)
 #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
 
@@ -75,9 +77,19 @@ static int ads117x_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#if defined(CONFIG_OF)
+static const struct of_device_id ads117x_dt_ids[] = {
+	{ .compatible = "ti,ads1174" },
+	{ .compatible = "ti,ads1178" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, ads117x_dt_ids);
+#endif
+
 static struct platform_driver ads117x_codec_driver = {
 	.driver = {
 			.name = "ads117x-codec",
+			.of_match_table = of_match_ptr(ads117x_dt_ids),
 	},
 
 	.probe = ads117x_probe,
-- 
cgit v0.10.2


From 5602a64318a4a6aad028141bb68e9bbd6bb5749a Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 10 Mar 2016 10:46:07 +0000
Subject: ASoC: wm_adsp: Use correct local length in error message

Fixes: 44029e9e1290 ("ASoC: wm_adsp: wm_coeff_{read|write}_control should use passed length")
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 36687db..ba9213c 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -787,7 +787,7 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
 	ret = regmap_raw_read(dsp->regmap, reg, scratch, len);
 	if (ret) {
 		adsp_err(dsp, "Failed to read %zu bytes from %x: %d\n",
-			 ctl->len, reg, ret);
+			 len, reg, ret);
 		kfree(scratch);
 		return ret;
 	}
-- 
cgit v0.10.2