From 6bd3c6f75e0f9baddbf1196a7e3fceabb50c7e3c Mon Sep 17 00:00:00 2001
From: Maciej S. Szmigiero
Date: Mon, 31 Aug 2015 17:07:12 +0200
Subject: ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device

In case of unknown DT compatible device the ASRC OF node
possibly acquired earlier by of_parse_phandle() has
to be put before returning from probe method.

Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl-asoc-card.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5aeb6ed4827e..96f55ae75c71 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
 	} else {
 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto asrc_fail;
 	}
 
 	/* Common settings for corresponding Freescale CPU DAI driver */
-- 
cgit v1.2.3


From d76f41982f2fc88492efd96c7c3178044f32e125 Mon Sep 17 00:00:00 2001
From: Anatol Pomozov
Date: Mon, 31 Aug 2015 08:24:23 -0700
Subject: ASoC: Document snd-soc-dummy-dai purpose

Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-utils.c | 9 +++++++++
 1 file changed, 9 insertions(+)

diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 362c69ac1d6c..53dd085d3ee2 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec;
 			SNDRV_PCM_FMTBIT_S32_LE | \
 			SNDRV_PCM_FMTBIT_U32_LE | \
 			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+/*
+ * The dummy CODEC is only meant to be used in situations where there is no
+ * actual hardware.
+ *
+ * If there is actual hardware even if it does not have a control bus
+ * the hardware will still have constraints like supported samplerates, etc.
+ * which should be modelled. And the data flow graph also should be modelled
+ * using DAPM.
+ */
 static struct snd_soc_dai_driver dummy_dai = {
 	.name = "snd-soc-dummy-dai",
 	.playback = {
-- 
cgit v1.2.3


From 51d2eeef1d958ef6834b24f548194f5acea0f499 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee
Date: Tue, 1 Sep 2015 11:14:05 +0530
Subject: ASoC: wm0010: fix memory leak

We were aborting if the kzalloc of img_swap fails but without freeing the
already allocated out. Similarly we were aborting if spi_sync fails
without releasing out and img_swap.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm0010.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 6560a66b3f35..9d370a45abe8 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -672,8 +672,10 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 		}
 
 		img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
-		if (!img_swap)
+		if (!img_swap) {
+			kfree(out);
 			goto abort;
+		}
 
 		/* We need to re-order for 0010 */
 		byte_swap_64((u64 *)&pll_rec, img_swap, len);
@@ -690,6 +692,8 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 		ret = spi_sync(spi, &m);
 		if (ret != 0) {
 			dev_err(codec->dev, "First PLL write failed: %d\n", ret);
+			kfree(img_swap);
+			kfree(out);
 			goto abort;
 		}
 
@@ -697,6 +701,8 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 		ret = spi_sync(spi, &m);
 		if (ret != 0) {
 			dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
+			kfree(img_swap);
+			kfree(out);
 			goto abort;
 		}
 
-- 
cgit v1.2.3


From 721b51fcf91898299d96f4b72cb9434cda29dce6 Mon Sep 17 00:00:00 2001
From: John Lin
Date: Wed, 9 Sep 2015 16:47:48 +0800
Subject: ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome
 platform

Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 7 +++++++
 1 file changed, 7 insertions(+)

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 610eacd34900..e620da164516 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3236,6 +3236,13 @@ static struct dmi_system_id dmi_platform_intel_braswell[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "Celes"),
 		},
 	},
+	{
+		.ident = "Google Ultima",
+		.callback = strago_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
+		},
+	},
 	{ }
 };
 
-- 
cgit v1.2.3


From f1ec5ec7a94ba8138f9cbdc1e9e3b03aee29c4df Mon Sep 17 00:00:00 2001
From: Liam Girdwood
Date: Wed, 9 Sep 2015 12:10:31 +0100
Subject: ASoC: intel: Fix SSP port configuration after RTD3 resume.

Currently the SSP port settings are being clobbered as part of the DSP
RTD3 restore logic. make sure we save the correct params and restore them
at resume. The FW sadly does not save SSP settings as part of the PM
context.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++--------
 1 file changed, 12 insertions(+), 8 deletions(-)

diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index f95f271aab0c..5bc98550c81d 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -302,6 +302,10 @@ struct sst_hsw {
 	struct sst_hsw_ipc_dx_reply dx;
 	void *dx_context;
 	dma_addr_t dx_context_paddr;
+	enum sst_hsw_device_id dx_dev;
+	enum sst_hsw_device_mclk dx_mclk;
+	enum sst_hsw_device_mode dx_mode;
+	u32 dx_clock_divider;
 
 	/* boot */
 	wait_queue_head_t boot_wait;
@@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
 
 	trace_ipc_request("set device config", dev);
 
-	config.ssp_interface = dev;
-	config.clock_frequency = mclk;
-	config.mode = mode;
-	config.clock_divider = clock_divider;
+	hsw->dx_dev = config.ssp_interface = dev;
+	hsw->dx_mclk = config.clock_frequency = mclk;
+	hsw->dx_mode = config.mode = mode;
+	hsw->dx_clock_divider = config.clock_divider = clock_divider;
 	if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER)
 		config.channels = 4;
 	else
@@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw)
 		return -EIO;
 	}
 
-	/* Set ADSP SSP port settings */
-	ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0,
-					SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
-					SST_HSW_DEVICE_CLOCK_MASTER, 9);
+	/* Set ADSP SSP port settings - sadly the FW does not store SSP port
+	   settings as part of the PM context. */
+	ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk,
+					hsw->dx_mode, hsw->dx_clock_divider);
 	if (ret < 0)
 		dev_err(dev, "error: SSP re-initialization failed\n");
 
-- 
cgit v1.2.3


From 3a0e27d84bb9abac5e39dc71706768a88c72cb71 Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen
Date: Thu, 10 Sep 2015 09:45:55 +0200
Subject: ASoC: sti: check return of of_property_read

Add check on of_property_read to return error when
DT required property is not defined.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sti/uniperif_player.c | 14 ++++++++++----
 sound/soc/sti/uniperif_reader.c |  6 +++++-
 2 files changed, 15 insertions(+), 5 deletions(-)

diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index f6eefe1b8f8f..843f037a317d 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev,
 	if (!info)
 		return -ENOMEM;
 
-	of_property_read_u32(pnode, "version", &player->ver);
-	if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
+	if (of_property_read_u32(pnode, "version", &player->ver) ||
+	    player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
 		dev_err(dev, "Unknown uniperipheral version ");
 		return -EINVAL;
 	}
@@ -998,10 +998,16 @@ static int uni_player_parse_dt(struct platform_device *pdev,
 	if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
 		info->underflow_enabled = 1;
 
-	of_property_read_u32(pnode, "uniperiph-id", &info->id);
+	if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) {
+		dev_err(dev, "uniperipheral id not defined");
+		return -EINVAL;
+	}
 
 	/* Read the device mode property */
-	of_property_read_string(pnode, "mode", &mode);
+	if (of_property_read_string(pnode, "mode", &mode)) {
+		dev_err(dev, "uniperipheral mode not defined");
+		return -EINVAL;
+	}
 
 	if (strcasecmp(mode, "hdmi") == 0)
 		info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI;
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index c502626f339b..f791239a3087 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
 	if (!info)
 		return -ENOMEM;
 
-	of_property_read_u32(node, "version", &reader->ver);
+	if (of_property_read_u32(node, "version", &reader->ver) ||
+	    reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
+		dev_err(&pdev->dev, "Unknown uniperipheral version ");
+		return -EINVAL;
+	}
 
 	/* Save the info structure */
 	reader->info = info;
-- 
cgit v1.2.3


From 75881df3fd7708f234c1e2573ade812eb5701708 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee
Date: Thu, 10 Sep 2015 18:01:44 +0530
Subject: ASoC: dapm: fix memory leak

Incase of an unknown event we were directly returning but we missed
freeing params.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f4bf21a5539b..ff8bda471b25 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3501,7 +3501,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 	default:
 		WARN(1, "Unknown event %d\n", event);
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
 out:
-- 
cgit v1.2.3


From f17b329b73a0393dc9d5fc5b4457189f92e5bbef Mon Sep 17 00:00:00 2001
From: Oder Chiou
Date: Thu, 10 Sep 2015 13:40:16 +0800
Subject: ASoC: rt5645: Remove incorrect settings

The patch removes the incorrect settings to avoid the pop sound in the
first playback with headphone after boot.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 6 ------
 1 file changed, 6 deletions(-)

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 4972bf3efa91..595ace9bcf58 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2829,10 +2829,6 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_sync(dapm);
 			rt5645->jack_type = SND_JACK_HEADPHONE;
 		}
-
-		snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
-		snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d);
-		snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001);
 	} else { /* jack out */
 		rt5645->jack_type = 0;
 
@@ -2880,8 +2876,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec,
 		rt5645->en_button_func = true;
 		regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1,
 				RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
-		regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
-				RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
 		regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
 				RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
 	}
-- 
cgit v1.2.3


From 3758ff5f3dab57cd768d54279962a2f6bbc17188 Mon Sep 17 00:00:00 2001
From: Zidan Wang
Date: Wed, 9 Sep 2015 19:29:10 +0800
Subject: ASoC: wm8960: correct the min gain value of some PGA

The min gain is the corresponding gain value when the register value is 0
instead of 1, just correct it.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8960.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 94c5c4681ce5..9c730490a02b 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -205,11 +205,11 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
 	return wm8960_set_deemph(codec);
 }
 
-static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
-static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
-static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
 
 static const struct snd_kcontrol_new wm8960_snd_controls[] = {
 SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
-- 
cgit v1.2.3


From 7e90f9b2b56669260e5f6f97974735d187f79b7d Mon Sep 17 00:00:00 2001
From: Zidan Wang
Date: Wed, 9 Sep 2015 19:29:11 +0800
Subject: ASoC: wm8960: correct gain value for input PGA and add microphone PGA

The input PGAs have a gain range from -17.25dB to +30dB in 0.75dB steps.

The boost stage can provide additional gain. For line inputs, -12dB to
+6dB gain is available on the boost mixer. For micphone inputs, it can
provide up to +29dB additional gain from the microphone PGA.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8960.c | 22 ++++++++++++++++------
 1 file changed, 16 insertions(+), 6 deletions(-)

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 9c730490a02b..70816f02ea8c 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -206,27 +206,37 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
 }
 
 static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1725, 75, 0);
 static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
-static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1);
+static const unsigned int micboost_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0),
+	2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0),
+};
 
 static const struct snd_kcontrol_new wm8960_snd_controls[] = {
 SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
-		 0, 63, 0, adc_tlv),
+		 0, 63, 0, inpga_tlv),
 SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
 	6, 1, 0),
 SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
 	7, 1, 0),
 
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
-	       WM8960_INBMIX1, 4, 7, 0, boost_tlv),
+	       WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
-	       WM8960_INBMIX1, 1, 7, 0, boost_tlv),
+	       WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
-	       WM8960_INBMIX2, 4, 7, 0, boost_tlv),
+	       WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
-	       WM8960_INBMIX2, 1, 7, 0, boost_tlv),
+	       WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
+		WM8960_RINPATH, 4, 4, 0, micboost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
+		WM8960_LINPATH, 4, 4, 0, micboost_tlv),
 
 SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
 		 0, 255, 0, dac_tlv),
-- 
cgit v1.2.3


From 723831927e8813b5b336d383174f686ad708bf10 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi
Date: Mon, 14 Sep 2015 16:06:48 +0300
Subject: ASoC: davinci-mcasp: Revise the FIFO threshold calculation

The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value
for the refining should meet this requirement as well.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index add6bb99661d..1260b315a96c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	u8 rx_ser = 0;
 	u8 slots = mcasp->tdm_slots;
 	u8 max_active_serializers = (channels + slots - 1) / slots;
-	int active_serializers, numevt, n;
+	int active_serializers, numevt;
 	u32 reg;
 	/* Default configuration */
 	if (mcasp->version < MCASP_VERSION_3)
@@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	 * The number of words for numevt need to be in steps of active
 	 * serializers.
 	 */
-	n = numevt % active_serializers;
-	if (n)
-		numevt += (active_serializers - n);
+	numevt = (numevt / active_serializers) * active_serializers;
+
 	while (period_words % numevt && numevt > 0)
 		numevt -= active_serializers;
 	if (numevt <= 0)
-- 
cgit v1.2.3


From e4fba9b5be12d577d2e2c19fdca6b0744c3f271e Mon Sep 17 00:00:00 2001
From: Koro Chen
Date: Mon, 14 Sep 2015 14:51:17 +0800
Subject: ASoC: mediatek: Increase periods_min in capture

In capture, there is chance that hw_ptr reported at IRQ is
a little smaller than period_size due to internal AFE buffer.
In the case of ping-pong buffer:

|xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------|
                            hw_ptr < period_size

This available buffer will not be read since its size is smaller than
avail_min (which is period_size by default), and read thread continues
to sleep. If the next hw_ptr is just a little larger than buffer_size,
overrun occurs. One more period can hold the possible unread buffer.

Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mtk-afe-pcm.c | 17 +++++++++++++++++
 1 file changed, 17 insertions(+)

diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index d190fe017559..f5baf3c38863 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -549,6 +549,23 @@ static int mtk_afe_dais_startup(struct snd_pcm_substream *substream,
 	memif->substream = substream;
 
 	snd_soc_set_runtime_hwparams(substream, &mtk_afe_hardware);
+
+	/*
+	 * Capture cannot use ping-pong buffer since hw_ptr at IRQ may be
+	 * smaller than period_size due to AFE's internal buffer.
+	 * This easily leads to overrun when avail_min is period_size.
+	 * One more period can hold the possible unread buffer.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		ret = snd_pcm_hw_constraint_minmax(runtime,
+						   SNDRV_PCM_HW_PARAM_PERIODS,
+						   3,
+						   mtk_afe_hardware.periods_max);
+		if (ret < 0) {
+			dev_err(afe->dev, "hw_constraint_minmax failed\n");
+			return ret;
+		}
+	}
 	ret = snd_pcm_hw_constraint_integer(runtime,
 					    SNDRV_PCM_HW_PARAM_PERIODS);
 	if (ret < 0)
-- 
cgit v1.2.3


From 42617869bf095c650e67aad4001cab4224e7fa98 Mon Sep 17 00:00:00 2001
From: Axel Lin
Date: Tue, 1 Sep 2015 12:31:24 +0800
Subject: ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM

devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/spear/Kconfig | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig
index 0a53053495f3..4fb91412ebec 100644
--- a/sound/soc/spear/Kconfig
+++ b/sound/soc/spear/Kconfig
@@ -1,6 +1,6 @@
 config SND_SPEAR_SOC
 	tristate
-	select SND_DMAENGINE_PCM
+	select SND_SOC_GENERIC_DMAENGINE_PCM
 
 config SND_SPEAR_SPDIF_OUT
 	tristate
-- 
cgit v1.2.3


From 921e54680aefe52f28d9ce9485edb1bfef4b92a8 Mon Sep 17 00:00:00 2001
From: Axel Lin
Date: Tue, 1 Sep 2015 14:50:15 +0800
Subject: ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning

Fix below build warning:
sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe':
sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable]

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/au1x/psc-i2s.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 38e853add96e..0bf9d62b91a0 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -296,7 +296,6 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 {
 	struct resource *iores, *dmares;
 	unsigned long sel;
-	int ret;
 	struct au1xpsc_audio_data *wd;
 
 	wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data),
-- 
cgit v1.2.3


From 295c3405a8bbd69ee8c8eb6580f30b0b8739b33a Mon Sep 17 00:00:00 2001
From: Jyri Sarha
Date: Wed, 9 Sep 2015 21:27:42 +0300
Subject: ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver

The TX and RX direction share the same bit clock and frame sync, so
the samplerate must be the same to both directions.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 1260b315a96c..2c5a2c5444ab 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1298,6 +1298,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 		.ops 		= &davinci_mcasp_dai_ops,
 
 		.symmetric_samplebits	= 1,
+		.symmetric_rates	= 1,
 	},
 	{
 		.name		= "davinci-mcasp.1",
-- 
cgit v1.2.3


From 3c8f7710c1c44fb650bc29b6ef78ed8b60cfaa28 Mon Sep 17 00:00:00 2001
From: Robert Jarzmik
Date: Tue, 15 Sep 2015 20:51:31 +0200
Subject: ASoC: fix broken pxa SoC support

The previous fix of pxa library support, which was introduced to fix the
library dependency, broke the previous SoC behavior, where a machine
code binding pxa2xx-ac97 with a coded relied on :
 - sound/soc/pxa/pxa2xx-ac97.c
 - sound/soc/codecs/XXX.c

For example, the mioa701_wm9713.c machine code is currently broken. The
"select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for
compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared
in sound/arm/Kconfig and sound/soc/pxa/Kconfig.

Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct
pxa2xx-ac97 compilation.

Fixes: 846172dfe33c ("ASoC: fix SND_PXA2XX_LIB Kconfig warning")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/arm/Kconfig     | 15 ++++++++-------
 sound/soc/pxa/Kconfig |  2 --
 2 files changed, 8 insertions(+), 9 deletions(-)

diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 885683a3b0bd..e0406211716b 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -9,6 +9,14 @@ menuconfig SND_ARM
 	  Drivers that are implemented on ASoC can be found in
 	  "ALSA for SoC audio support" section.
 
+config SND_PXA2XX_LIB
+	tristate
+	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
+	select SND_DMAENGINE_PCM
+
+config SND_PXA2XX_LIB_AC97
+	bool
+
 if SND_ARM
 
 config SND_ARMAACI
@@ -21,13 +29,6 @@ config SND_PXA2XX_PCM
 	tristate
 	select SND_PCM
 
-config SND_PXA2XX_LIB
-	tristate
-	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
-
-config SND_PXA2XX_LIB_AC97
-	bool
-
 config SND_PXA2XX_AC97
 	tristate "AC97 driver for the Intel PXA2xx chip"
 	depends on ARCH_PXA
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 39cea80846c3..f2bf8661dd21 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,7 +1,6 @@
 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
 	depends on ARCH_PXA
-	select SND_ARM
 	select SND_PXA2XX_LIB
 	help
 	  Say Y or M if you want to add support for codecs attached to
@@ -25,7 +24,6 @@ config SND_PXA2XX_AC97
 config SND_PXA2XX_SOC_AC97
 	tristate
 	select AC97_BUS
-	select SND_ARM
 	select SND_PXA2XX_LIB_AC97
 	select SND_SOC_AC97_BUS
 
-- 
cgit v1.2.3


From ab1fffe3a73c694d698645451ba61255ec4ba5e6 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi
Date: Fri, 18 Sep 2015 15:02:50 +0300
Subject: ASoC: davinci-mcasp: Fix devm_kasprintf format string

The '\n' at the end of the format string is not needed. It adds an extra
line break when doing
cat /proc/interrupts

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 2c5a2c5444ab..7d45d98a861f 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1685,7 +1685,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq_byname(pdev, "common");
 	if (irq >= 0) {
-		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n",
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
 					  dev_name(&pdev->dev));
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_common_irq_handler,
@@ -1702,7 +1702,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq_byname(pdev, "rx");
 	if (irq >= 0) {
-		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
 					  dev_name(&pdev->dev));
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_rx_irq_handler,
@@ -1717,7 +1717,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq_byname(pdev, "tx");
 	if (irq >= 0) {
-		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n",
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
 					  dev_name(&pdev->dev));
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_tx_irq_handler,
-- 
cgit v1.2.3


From ee92cfb030c16ddb01f6543968f13bcb61ed9da5 Mon Sep 17 00:00:00 2001
From: Zidan Wang
Date: Fri, 18 Sep 2015 17:19:25 +0800
Subject: ASoC: wm8962: remove 64k sample rate support

wm8962 can't support 64k sample rate. When playing a 64KHz wave file,
'Unsupported rate 64000Hz' will be prompted.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8962.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index b4eb975da981..293e47a6ff59 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2944,7 +2944,8 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute)
 				   WM8962_DAC_MUTE, val);
 }
 
-#define WM8962_RATES SNDRV_PCM_RATE_8000_96000
+#define WM8962_RATES (SNDRV_PCM_RATE_8000_48000 |\
+		SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-- 
cgit v1.2.3


From 8524bb0c7ac688a3cd6ba12dae6104c54d0566b9 Mon Sep 17 00:00:00 2001
From: Zidan Wang
Date: Fri, 18 Sep 2015 17:19:43 +0800
Subject: ASoC: wm8960: correct the max register value of mic boost pga

the max register value of mic boost pga should be 3.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8960.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 70816f02ea8c..d9b43b00244e 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -234,9 +234,9 @@ SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
 	       WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
-		WM8960_RINPATH, 4, 4, 0, micboost_tlv),
+		WM8960_RINPATH, 4, 3, 0, micboost_tlv),
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
-		WM8960_LINPATH, 4, 4, 0, micboost_tlv),
+		WM8960_LINPATH, 4, 3, 0, micboost_tlv),
 
 SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
 		 0, 255, 0, dac_tlv),
-- 
cgit v1.2.3


From 2ace47be5a315def8f493ca77aa59c077ade30a1 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee
Date: Fri, 18 Sep 2015 16:02:20 +0530
Subject: ASoC: wm0010: fix memory leak

We have requested for the firmware but we have missed releasing it both
on success and on error path.
While checking the code it turned out that the requested firmware is not
even used. More over the same firmware is being loaded by
wm0010_stage2_load().

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm0010.c | 9 ---------
 1 file changed, 9 deletions(-)

diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 9d370a45abe8..7a1bc40538ca 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -577,7 +577,6 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 	struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
 	unsigned long flags;
 	int ret;
-	const struct firmware *fw;
 	struct spi_message m;
 	struct spi_transfer t;
 	struct dfw_pllrec pll_rec;
@@ -623,14 +622,6 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 	wm0010->state = WM0010_OUT_OF_RESET;
 	spin_unlock_irqrestore(&wm0010->irq_lock, flags);
 
-	/* First the bootloader */
-	ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to request stage2 loader: %d\n",
-			ret);
-		goto abort;
-	}
-
 	if (!wait_for_completion_timeout(&wm0010->boot_completion,
 					 msecs_to_jiffies(20)))
 		dev_err(codec->dev, "Failed to get interrupt from DSP\n");
-- 
cgit v1.2.3


From f072f91aa7517386344476813ca0799e08fd0c35 Mon Sep 17 00:00:00 2001
From: Sudip Mukherjee
Date: Fri, 18 Sep 2015 16:02:21 +0530
Subject: ASoC: wm0010: fix error path

Fix the error path so that we can free the allocated memory on the error
path instead of releasing them individually on each error.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm0010.c | 22 ++++++++++------------
 1 file changed, 10 insertions(+), 12 deletions(-)

diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 7a1bc40538ca..76b2f88d9d9b 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -663,10 +663,8 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 		}
 
 		img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
-		if (!img_swap) {
-			kfree(out);
-			goto abort;
-		}
+		if (!img_swap)
+			goto abort_out;
 
 		/* We need to re-order for 0010 */
 		byte_swap_64((u64 *)&pll_rec, img_swap, len);
@@ -681,20 +679,16 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 		spi_message_add_tail(&t, &m);
 
 		ret = spi_sync(spi, &m);
-		if (ret != 0) {
+		if (ret) {
 			dev_err(codec->dev, "First PLL write failed: %d\n", ret);
-			kfree(img_swap);
-			kfree(out);
-			goto abort;
+			goto abort_swap;
 		}
 
 		/* Use a second send of the message to get the return status */
 		ret = spi_sync(spi, &m);
-		if (ret != 0) {
+		if (ret) {
 			dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
-			kfree(img_swap);
-			kfree(out);
-			goto abort;
+			goto abort_swap;
 		}
 
 		p = (u32 *)out;
@@ -727,6 +721,10 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 
 	return 0;
 
+abort_swap:
+	kfree(img_swap);
+abort_out:
+	kfree(out);
 abort:
 	/* Put the chip back into reset */
 	wm0010_halt(codec);
-- 
cgit v1.2.3


From 5b64c173cdea21105eb4794487b3d593f0a2e6c3 Mon Sep 17 00:00:00 2001
From: Adam Thomson
Date: Wed, 16 Sep 2015 10:13:19 +0100
Subject: ASoC: fsl_ssi: Fix checking of dai format for AC97 mode

Current code incorrectly treats dai format for AC97 as bit mask
whereas it's actually an integer value. This causes DAI formats
other than AC97 (e.g. DSP_B) to trigger AC97 related code,
which is incorrect and breaks functionality. This patch fixes
the code to correctly compare values to determine AC97 or not.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_ssi.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 8ec6fb208ea0..37c5cd4d0e59 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
 
 static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private)
 {
-	return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97);
+	return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+		SND_SOC_DAIFMT_AC97;
 }
 
 static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
@@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 				CCSR_SSI_SCR_TCH_EN);
 	}
 
-	if (fmt & SND_SOC_DAIFMT_AC97)
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97)
 		fsl_ssi_setup_ac97(ssi_private);
 
 	return 0;
-- 
cgit v1.2.3


From f0e03dbd2d61d991bdd2d76b4e84681fe3077176 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald
Date: Tue, 22 Sep 2015 15:30:33 +0100
Subject: MAINTAINERS: Update website and git repo for Wolfson Microelectronics

Support for Wolfson Microelectronics devices is now part of Cirrus Logic
and the relevant parts of the old opensource.wolfsonmicro.com site have
moved to the Cirrus Logic GitHub area.

This patch updates the website and git repo links, and also removes an
obsolete website link for the voltage and current drivers.

Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 MAINTAINERS | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

diff --git a/MAINTAINERS b/MAINTAINERS
index 7ba7ab749c85..dffea210e439 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -11239,7 +11239,6 @@ VOLTAGE AND CURRENT REGULATOR FRAMEWORK
 M:	Liam Girdwood <lgirdwood@gmail.com>
 M:	Mark Brown <broonie@kernel.org>
 L:	linux-kernel@vger.kernel.org
-W:	http://opensource.wolfsonmicro.com/node/15
 W:	http://www.slimlogic.co.uk/?p=48
 T:	git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator.git
 S:	Supported
@@ -11368,17 +11367,15 @@ WM97XX TOUCHSCREEN DRIVERS
 M:	Mark Brown <broonie@kernel.org>
 M:	Liam Girdwood <lrg@slimlogic.co.uk>
 L:	linux-input@vger.kernel.org
-T:	git git://opensource.wolfsonmicro.com/linux-2.6-touch
-W:	http://opensource.wolfsonmicro.com/node/7
+W:	https://github.com/CirrusLogic/linux-drivers/wiki
 S:	Supported
 F:	drivers/input/touchscreen/*wm97*
 F:	include/linux/wm97xx.h
 
 WOLFSON MICROELECTRONICS DRIVERS
 L:	patches@opensource.wolfsonmicro.com
-T:	git git://opensource.wolfsonmicro.com/linux-2.6-asoc
-T:	git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
-W:	http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
+T:	git https://github.com/CirrusLogic/linux-drivers.git
+W:	https://github.com/CirrusLogic/linux-drivers/wiki
 S:	Supported
 F:	Documentation/hwmon/wm83??
 F:	arch/arm/mach-s3c64xx/mach-crag6410*
-- 
cgit v1.2.3


From 8811191fdf7ed02ee07cb8469428158572d355a2 Mon Sep 17 00:00:00 2001
From: Robert Jarzmik
Date: Tue, 22 Sep 2015 21:20:22 +0200
Subject: ASoC: pxa: pxa2xx-ac97: fix dma requestor lines

PCM receive and transmit DMA requestor lines were reverted, breaking the
PCM playback interface for PXA platforms using the sound/soc/ variant
instead of the sound/arm variant.

The commit below shows the inversion in the requestor lines.

Fixes: d65a14587a9b ("ASoC: pxa: use snd_dmaengine_dai_dma_data")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
---
 sound/soc/pxa/pxa2xx-ac97.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1f6054650991..9e4b04e0fbd1 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_cold_reset,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
-- 
cgit v1.2.3


From 21cb13e72b02dbbb5a02477d4dd46bc2bc1cfd08 Mon Sep 17 00:00:00 2001
From: Oder Chiou
Date: Wed, 23 Sep 2015 14:35:28 +0800
Subject: ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the
 weird sound in runtime of power up

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index dbc1d76d9d4e..25c34fbae307 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -732,14 +732,14 @@ static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = {
 static const struct snd_kcontrol_new rt5645_dac_l_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_ADCMIX_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_DAC1_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5645_dac_r_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_ADCMIX_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_DAC1_R_SFT, 1, 1),
 };
 
-- 
cgit v1.2.3


From 4f4794124e0421b080a0f1f5f1207636ba55eb85 Mon Sep 17 00:00:00 2001
From: Oder Chiou
Date: Wed, 23 Sep 2015 14:35:29 +0800
Subject: ASoC: rt5645: Increase the delay time to remove the pop sound

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 25c34fbae307..61384ee989fc 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -1381,7 +1381,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
 				regmap_write(rt5645->regmap, RT5645_PR_BASE +
 					RT5645_MAMP_INT_REG2, 0xfc00);
 				snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
-				mdelay(5);
+				msleep(40);
 				rt5645->hp_on = true;
 			} else {
 				/* depop parameters */
-- 
cgit v1.2.3


From fce97b4d70ad632dd9c6058622492501377bbaaa Mon Sep 17 00:00:00 2001
From: Oder Chiou
Date: Wed, 23 Sep 2015 14:35:30 +0800
Subject: ASoC: rt5645: Prevent the pop sound in case of playback and the jack
 is plugging

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 61384ee989fc..268a28bd1df4 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2832,6 +2832,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 	} else { /* jack out */
 		rt5645->jack_type = 0;
 
+		regmap_update_bits(rt5645->regmap, RT5645_HP_VOL,
+			RT5645_L_MUTE | RT5645_R_MUTE,
+			RT5645_L_MUTE | RT5645_R_MUTE);
 		regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2,
 			RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD);
 		regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1,
-- 
cgit v1.2.3


From 83510441bc08bee201c0ded9d81da6dfd008d69a Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Thu, 24 Sep 2015 11:00:18 +0200
Subject: ALSA: hda/tegra - async probe for avoiding module loading deadlock

The Tegra HD-audio controller driver causes deadlocks when loaded as a
module since the driver invokes request_module() at binding with the
codec driver.  This patch works around it by deferring the probe in a
work like Intel HD-audio controller driver does.  Although hovering
the codec probe stuff into udev would be a better solution, it may
cause other regressions, so let's try this band-aid fix until the more
proper solution gets landed.

Reported-by: Thierry Reding <treding@nvidia.com>
Tested-by: Thierry Reding <treding@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_tegra.c | 30 +++++++++++++++++++++++++-----
 1 file changed, 25 insertions(+), 5 deletions(-)

diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 477742cb70a2..58c0aad37284 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -73,6 +73,7 @@ struct hda_tegra {
 	struct clk *hda2codec_2x_clk;
 	struct clk *hda2hdmi_clk;
 	void __iomem *regs;
+	struct work_struct probe_work;
 };
 
 #ifdef CONFIG_PM
@@ -294,7 +295,9 @@ static int hda_tegra_dev_disconnect(struct snd_device *device)
 static int hda_tegra_dev_free(struct snd_device *device)
 {
 	struct azx *chip = device->device_data;
+	struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
 
+	cancel_work_sync(&hda->probe_work);
 	if (azx_bus(chip)->chip_init) {
 		azx_stop_all_streams(chip);
 		azx_stop_chip(chip);
@@ -426,6 +429,9 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev)
 /*
  * constructor
  */
+
+static void hda_tegra_probe_work(struct work_struct *work);
+
 static int hda_tegra_create(struct snd_card *card,
 			    unsigned int driver_caps,
 			    struct hda_tegra *hda)
@@ -452,6 +458,8 @@ static int hda_tegra_create(struct snd_card *card,
 	chip->single_cmd = false;
 	chip->snoop = true;
 
+	INIT_WORK(&hda->probe_work, hda_tegra_probe_work);
+
 	err = azx_bus_init(chip, NULL, &hda_tegra_io_ops);
 	if (err < 0)
 		return err;
@@ -499,6 +507,21 @@ static int hda_tegra_probe(struct platform_device *pdev)
 	card->private_data = chip;
 
 	dev_set_drvdata(&pdev->dev, card);
+	schedule_work(&hda->probe_work);
+
+	return 0;
+
+out_free:
+	snd_card_free(card);
+	return err;
+}
+
+static void hda_tegra_probe_work(struct work_struct *work)
+{
+	struct hda_tegra *hda = container_of(work, struct hda_tegra, probe_work);
+	struct azx *chip = &hda->chip;
+	struct platform_device *pdev = to_platform_device(hda->dev);
+	int err;
 
 	err = hda_tegra_first_init(chip, pdev);
 	if (err < 0)
@@ -520,11 +543,8 @@ static int hda_tegra_probe(struct platform_device *pdev)
 	chip->running = 1;
 	snd_hda_set_power_save(&chip->bus, power_save * 1000);
 
-	return 0;
-
-out_free:
-	snd_card_free(card);
-	return err;
+ out_free:
+	return; /* no error return from async probe */
 }
 
 static int hda_tegra_remove(struct platform_device *pdev)
-- 
cgit v1.2.3


From 7f57d803ee03730d570dc59a9e3e4842b58dd5cc Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Thu, 24 Sep 2015 17:36:51 +0200
Subject: ALSA: hda - Disable power_save_node for Thinkpads

Lenovo Thinkpads with recent Realtek codecs seem suffering from click
noises at power transition since the introduction of widget power
saving in 4.1 kernel.  Although this might be solved by some delays in
appropriate points, as a quick workaround, just disable the
power_save_node feature for now.  The gain it gives is relatively
small, and this makes the situation back to pre 4.1 time.

This patch ended up with a bit more code changes than usual because
the existing fixup for Thinkpads is highly chained.  Instead of adding
yet another chain, combine a few of them into a single fixup entry, as
a gratis cleanup.

Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 31 +++++++++++++++++++------------
 1 file changed, 19 insertions(+), 12 deletions(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a75b5611d1e4..afec6dc9f91f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4188,6 +4188,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
 	}
 }
 
+/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */
+static void alc_fixup_tpt440_dock(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	static const struct hda_pintbl pincfgs[] = {
+		{ 0x16, 0x21211010 }, /* dock headphone */
+		{ 0x19, 0x21a11010 }, /* dock mic */
+		{ }
+	};
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
+		codec->power_save_node = 0; /* avoid click noises */
+		snd_hda_apply_pincfgs(codec, pincfgs);
+	}
+}
+
 static void alc_shutup_dell_xps13(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4562,7 +4580,6 @@ enum {
 	ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
 	ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC292_FIXUP_TPT440_DOCK,
-	ALC292_FIXUP_TPT440_DOCK2,
 	ALC283_FIXUP_BXBT2807_MIC,
 	ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
 	ALC282_FIXUP_ASPIRE_V5_PINS,
@@ -5029,17 +5046,7 @@ static const struct hda_fixup alc269_fixups[] = {
 	},
 	[ALC292_FIXUP_TPT440_DOCK] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
-		.chained = true,
-		.chain_id = ALC292_FIXUP_TPT440_DOCK2
-	},
-	[ALC292_FIXUP_TPT440_DOCK2] = {
-		.type = HDA_FIXUP_PINS,
-		.v.pins = (const struct hda_pintbl[]) {
-			{ 0x16, 0x21211010 }, /* dock headphone */
-			{ 0x19, 0x21a11010 }, /* dock mic */
-			{ }
-		},
+		.v.func = alc_fixup_tpt440_dock,
 		.chained = true,
 		.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
 	},
-- 
cgit v1.2.3