Merge tag 'sound-4.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This ended up with a larger set of fixes than wished, unfortunately.

  As diffstat shows, the majority of changes are for various ASoC
  drivers (Realtek, Wolfson codec drivers, etc), in addition to a couple
  of HD-audio regression fixes.  All these are reasonably small and
  nothing to scare much"

* tag 'sound-4.3-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits)
  ALSA: hda - Disable power_save_node for Thinkpads
  ALSA: hda/tegra - async probe for avoiding module loading deadlock
  ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging
  ASoC: rt5645: Increase the delay time to remove the pop sound
  ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up
  ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
  MAINTAINERS: Update website and git repo for Wolfson Microelectronics
  ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
  ASoC: wm0010: fix error path
  ASoC: wm0010: fix memory leak
  ASoC: wm8960: correct the max register value of mic boost pga
  ASoC: wm8962: remove 64k sample rate support
  ASoC: davinci-mcasp: Fix devm_kasprintf format string
  ASoC: fix broken pxa SoC support
  ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver
  ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning
  ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
  ASoC: mediatek: Increase periods_min in capture
  ASoC: davinci-mcasp: Revise the FIFO threshold calculation
  ASoC: wm8960: correct gain value for input PGA and add microphone PGA
  ...

MAINTAINERS

@@ -11249,7 +11249,6 @@ 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
@@ -11378,17 +11377,15 @@ 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*

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

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)

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
 	},

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),

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),
 };
 
@@ -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 */
@@ -2829,13 +2829,12 @@ 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;
 
+		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,
@@ -2880,8 +2879,6 @@ 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);
 	}
@@ -3205,6 +3202,13 @@ static const 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"),
+		},
+	},
 	{ }
 };
 

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");
@@ -673,7 +664,7 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 
 		img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
 		if (!img_swap)
-			goto abort;
+			goto abort_out;
 
 		/* We need to re-order for 0010 */
 		byte_swap_64((u64 *)&pll_rec, img_swap, len);
@@ -688,16 +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);
-			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);
-			goto abort;
+			goto abort_swap;
 		}
 
 		p = (u32 *)out;
@@ -730,6 +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);

sound/soc/codecs/wm8960.c

@@ -211,28 +211,38 @@ 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(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, -1200, 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, 3, 0, micboost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
+		WM8960_LINPATH, 4, 3, 0, micboost_tlv),
 
 SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
 		 0, 255, 0, dac_tlv),

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)

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)
@@ -1299,6 +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",
@@ -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,

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 */

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;

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");
 

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)

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
 

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,

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:

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 = {

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

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;

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;