Merge tag 'sound-3.15-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Unfortunately this update became bigger than previous pull requests,
  which is almost a pattern in rc5-6.  But, the only obvious big changes
  are for the new Intel DSP ASoC drivers, so the impact must be fairly
  limited.

  Other than that, usual small fixes in various fields: HD-audio, ASoC
  core and ASoC fsl and codec drivers"

* tag 'sound-3.15-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (26 commits)
  ALSA: sb_mixer: missing return statement
  ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile
  ASoC: Intel: Fix Baytrail SST DSP firmware loading
  ALSA: hda - mask buggy stream DMA0 for Broadwell display controller
  ALSA: hda - Add new GPU codec ID to snd-hda
  ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params()
  ASoC: fsl_esai: Only bypass sck_div for EXTAL source
  ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP
  ASoC: dapm: Fix SUSPEND -> OFF bias sequence
  ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets()
  ASoC: pcm: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND
  ALSA: hda - add headset mic detect quirks for three Dell laptops
  ASoC: Update Cirrus Logic CODEC maintainers.
  ASoC: Intel: Fix block offset calculations.
  ASoC: Intel: Fix check for pdata usage before dereference.
  ASoC: Intel: Fix stream position pointer.
  ASoC: Intel: Fix allow hw_params to be called more than once.
  ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.
  ASoC: Intel: Fix Haswell/Broadwell DSP page table creation.
  ASoC: Intel: Fix allocated block list usage when adding blocks.
  ...

24 files changed, 190 insertions, 92 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index 2e4fe5450fea..6846c7c622e3 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -2246,12 +2246,6 @@ L:	linux-usb@vger.kernel.org
 S:	Maintained
 F:	drivers/usb/host/ohci-ep93xx.c
 
-CIRRUS LOGIC CS4270 SOUND DRIVER
-M:	Timur Tabi <timur@tabi.org>
-L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
-S:	Odd Fixes
-F:	sound/soc/codecs/cs4270*
-
 CIRRUS LOGIC AUDIO CODEC DRIVERS
 M:	Brian Austin <brian.austin@cirrus.com>
 M:	Paul Handrigan <Paul.Handrigan@cirrus.com>
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 6496822c1808..1ff78ec9f0ac 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -818,12 +818,14 @@ int snd_sbmixer_new(struct snd_sb *chip)
 			return err;
 		break;
 	case SB_HW_DT019X:
-		if ((err = snd_sbmixer_init(chip,
-					    snd_dt019x_controls,
-					    ARRAY_SIZE(snd_dt019x_controls),
-					    snd_dt019x_init_values,
-					    ARRAY_SIZE(snd_dt019x_init_values),
-					    "DT019X")) < 0)
+		err = snd_sbmixer_init(chip,
+				       snd_dt019x_controls,
+				       ARRAY_SIZE(snd_dt019x_controls),
+				       snd_dt019x_init_values,
+				       ARRAY_SIZE(snd_dt019x_init_values),
+				       "DT019X");
+		if (err < 0)
+			return err;
 		break;
 	default:
 		strcpy(card->mixername, "???");
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b540ad71eb0d..2c54629d62d1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1367,6 +1367,12 @@ static int azx_first_init(struct azx *chip)
 	/* initialize streams */
 	azx_init_stream(chip);
 
+	/* workaround for Broadwell HDMI: the first stream is broken,
+	 * so mask it by keeping it as if opened
+	 */
+	if (pci->vendor == 0x8086 && pci->device == 0x160c)
+		chip->azx_dev[0].opened = 1;
+
 	/* initialize chip */
 	azx_init_pci(chip);
 	azx_init_chip(chip, (probe_only[dev] & 2) == 0);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 1edbb9c47c2d..b4218a19df22 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3332,6 +3332,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x10de0051, .name = "GPU 51 HDMI/DP",	.patch = patch_nvhdmi },
 { .id = 0x10de0060, .name = "GPU 60 HDMI/DP",	.patch = patch_nvhdmi },
 { .id = 0x10de0067, .name = "MCP67 HDMI",	.patch = patch_nvhdmi_2ch },
+{ .id = 0x10de0071, .name = "GPU 71 HDMI/DP",	.patch = patch_nvhdmi },
 { .id = 0x10de8001, .name = "MCP73 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x11069f80, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
 { .id = 0x11069f81, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
@@ -3387,6 +3388,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0044");
 MODULE_ALIAS("snd-hda-codec-id:10de0051");
 MODULE_ALIAS("snd-hda-codec-id:10de0060");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
+MODULE_ALIAS("snd-hda-codec-id:10de0071");
 MODULE_ALIAS("snd-hda-codec-id:10de8001");
 MODULE_ALIAS("snd-hda-codec-id:11069f80");
 MODULE_ALIAS("snd-hda-codec-id:11069f81");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5f7c765391f1..49e884fb3e5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4616,6 +4616,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0657, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x065c, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -4624,6 +4625,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0674, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x067e, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0680, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0684, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index fa158cfe9b32..d1929de641e2 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -376,7 +376,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
 		reg = AIC31XX_ADCFLAG;
 		break;
 	default:
-		dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
+		dev_err(w->codec->dev, "Unknown widget '%s' calling %s\n",
 			w->name, __func__);
 		return -EINVAL;
 	}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 5522d2566c67..ecd26dd2e442 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -154,6 +154,7 @@ static struct reg_default wm8962_reg[] = {
 	{ 40, 0x0000 },   /* R40    - SPKOUTL volume */
 	{ 41, 0x0000 },   /* R41    - SPKOUTR volume */
 
+	{ 49, 0x0010 },   /* R49    - Class D Control 1 */
 	{ 51, 0x0003 },   /* R51    - Class D Control 2 */
 
 	{ 56, 0x0506 },   /* R56    - Clocking 4 */
@@ -795,7 +796,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg)
 	case WM8962_ALC2:
 	case WM8962_THERMAL_SHUTDOWN_STATUS:
 	case WM8962_ADDITIONAL_CONTROL_4:
-	case WM8962_CLASS_D_CONTROL_1:
 	case WM8962_DC_SERVO_6:
 	case WM8962_INTERRUPT_STATUS_1:
 	case WM8962_INTERRUPT_STATUS_2:
@@ -2929,13 +2929,22 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 static int wm8962_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	int val;
+	int val, ret;
 
 	if (mute)
-		val = WM8962_DAC_MUTE;
+		val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT;
 	else
 		val = 0;
 
+	/**
+	 * The DAC mute bit is mirrored in two registers, update both to keep
+	 * the register cache consistent.
+	 */
+	ret = snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_1,
+				  WM8962_DAC_MUTE_ALT, val);
+	if (ret < 0)
+		return ret;
+
 	return snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1,
 				   WM8962_DAC_MUTE, val);
 }
diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h
index a1a5d5294c19..910aafd09d21 100644
--- a/sound/soc/codecs/wm8962.h
+++ b/sound/soc/codecs/wm8962.h
@@ -1954,6 +1954,10 @@
 #define WM8962_SPKOUTL_ENA_MASK                 0x0040  /* SPKOUTL_ENA */
 #define WM8962_SPKOUTL_ENA_SHIFT                     6  /* SPKOUTL_ENA */
 #define WM8962_SPKOUTL_ENA_WIDTH                     1  /* SPKOUTL_ENA */
+#define WM8962_DAC_MUTE_ALT                     0x0010  /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_MASK                0x0010  /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_SHIFT                    4  /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_WIDTH                    1  /* DAC_MUTE */
 #define WM8962_SPKOUTL_PGA_MUTE                 0x0002  /* SPKOUTL_PGA_MUTE */
 #define WM8962_SPKOUTL_PGA_MUTE_MASK            0x0002  /* SPKOUTL_PGA_MUTE */
 #define WM8962_SPKOUTL_PGA_MUTE_SHIFT                1  /* SPKOUTL_PGA_MUTE */
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c8e5db1414d7..496ce2eb2f1f 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -258,10 +258,16 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 		return -EINVAL;
 	}
 
-	if (ratio == 1) {
+	/* Only EXTAL source can be output directly without using PSR and PM */
+	if (ratio == 1 && clksrc == esai_priv->extalclk) {
 		/* Bypass all the dividers if not being needed */
 		ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO;
 		goto out;
+	} else if (ratio < 2) {
+		/* The ratio should be no less than 2 if using other sources */
+		dev_err(dai->dev, "failed to derive required HCK%c rate\n",
+				tx ? 'T' : 'R');
+		return -EINVAL;
 	}
 
 	ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0);
@@ -307,7 +313,8 @@ static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
 		return -EINVAL;
 	}
 
-	if (esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) {
+	/* The ratio should be contented by FP alone if bypassing PM and PSR */
+	if (!esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) {
 		dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n");
 		return -EINVAL;
 	}
@@ -454,12 +461,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
 	}
 
 	if (!dai->active) {
-		/* Reset Port C */
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
-				   ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
-				   ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
-
 		/* Set synchronous mode */
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR,
 				   ESAI_SAICR_SYNC, esai_priv->synchronous ?
@@ -519,6 +520,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
 
+	/* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
+			   ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
+			   ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
 	return 0;
 }
 
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index ac869931d7f1..267717aa96c1 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -145,7 +145,7 @@ static const struct file_operations audmux_debugfs_fops = {
 	.llseek = default_llseek,
 };
 
-static void __init audmux_debugfs_init(void)
+static void audmux_debugfs_init(void)
 {
 	int i;
 	char buf[20];
diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
index 5d06eecb6198..18aee77f8d4a 100644
--- a/sound/soc/intel/sst-acpi.c
+++ b/sound/soc/intel/sst-acpi.c
@@ -138,6 +138,7 @@ static int sst_acpi_probe(struct platform_device *pdev)
 
 	sst_pdata = &sst_acpi->sst_pdata;
 	sst_pdata->id = desc->sst_id;
+	sst_pdata->dma_dev = dev;
 	sst_acpi->desc = desc;
 	sst_acpi->mach = mach;
 
diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c
index a50bf7fc0e3a..adf0aca5aca6 100644
--- a/sound/soc/intel/sst-baytrail-dsp.c
+++ b/sound/soc/intel/sst-baytrail-dsp.c
@@ -324,7 +324,7 @@ static int sst_byt_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 	memcpy_toio(sst->addr.lpe + SST_BYT_MAILBOX_OFFSET,
 	       &pdata->fw_base, sizeof(u32));
 
-	ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32));
+	ret = dma_coerce_mask_and_coherent(sst->dma_dev, DMA_BIT_MASK(32));
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index d0eaeee21be4..0d31dbbf4806 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -542,16 +542,20 @@ struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id,
 	void *data)
 {
 	struct sst_byt_stream *stream;
+	struct sst_dsp *sst = byt->dsp;
+	unsigned long flags;
 
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (stream == NULL)
 		return NULL;
 
+	spin_lock_irqsave(&sst->spinlock, flags);
 	list_add(&stream->node, &byt->stream_list);
 	stream->notify_position = notify_position;
 	stream->pdata = data;
 	stream->byt = byt;
 	stream->str_id = id;
+	spin_unlock_irqrestore(&sst->spinlock, flags);
 
 	return stream;
 }
@@ -630,6 +634,8 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream)
 {
 	u64 header;
 	int ret = 0;
+	struct sst_dsp *sst = byt->dsp;
+	unsigned long flags;
 
 	if (!stream->commited)
 		goto out;
@@ -644,8 +650,10 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream)
 
 	stream->commited = false;
 out:
+	spin_lock_irqsave(&sst->spinlock, flags);
 	list_del(&stream->node);
 	kfree(stream);
+	spin_unlock_irqrestore(&sst->spinlock, flags);
 
 	return ret;
 }
diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h
index 30ca14a6a835..401213455497 100644
--- a/sound/soc/intel/sst-dsp-priv.h
+++ b/sound/soc/intel/sst-dsp-priv.h
@@ -228,6 +228,7 @@ struct sst_dsp {
 	spinlock_t spinlock;	/* IPC locking */
 	struct mutex mutex;	/* DSP FW lock */
 	struct device *dev;
+	struct device *dma_dev;
 	void *thread_context;
 	int irq;
 	u32 id;
diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c
index 0c129fd85ecf..0b715b20a2d7 100644
--- a/sound/soc/intel/sst-dsp.c
+++ b/sound/soc/intel/sst-dsp.c
@@ -337,6 +337,7 @@ struct sst_dsp *sst_dsp_new(struct device *dev,
 	spin_lock_init(&sst->spinlock);
 	mutex_init(&sst->mutex);
 	sst->dev = dev;
+	sst->dma_dev = pdata->dma_dev;
 	sst->thread_context = sst_dev->thread_context;
 	sst->sst_dev = sst_dev;
 	sst->id = pdata->id;
diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h
index 74052b59485c..e44423be66c4 100644
--- a/sound/soc/intel/sst-dsp.h
+++ b/sound/soc/intel/sst-dsp.h
@@ -169,6 +169,7 @@ struct sst_pdata {
 	u32 dma_base;
 	u32 dma_size;
 	int dma_engine;
+	struct device *dma_dev;
 
 	/* DSP */
 	u32 id;
diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c
index f7687107cf7f..928f228c38e7 100644
--- a/sound/soc/intel/sst-firmware.c
+++ b/sound/soc/intel/sst-firmware.c
@@ -57,14 +57,8 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
 	sst_fw->private = private;
 	sst_fw->size = fw->size;
 
-	err = dma_coerce_mask_and_coherent(dsp->dev, DMA_BIT_MASK(32));
-	if (err < 0) {
-		kfree(sst_fw);
-		return NULL;
-	}
-
 	/* allocate DMA buffer to store FW data */
-	sst_fw->dma_buf = dma_alloc_coherent(dsp->dev, sst_fw->size,
+	sst_fw->dma_buf = dma_alloc_coherent(dsp->dma_dev, sst_fw->size,
 				&sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL);
 	if (!sst_fw->dma_buf) {
 		dev_err(dsp->dev, "error: DMA alloc failed\n");
@@ -106,7 +100,7 @@ void sst_fw_free(struct sst_fw *sst_fw)
 	list_del(&sst_fw->list);
 	mutex_unlock(&dsp->mutex);
 
-	dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf,
+	dma_free_coherent(dsp->dma_dev, sst_fw->size, sst_fw->dma_buf,
 			sst_fw->dmable_fw_paddr);
 	kfree(sst_fw);
 }
@@ -202,6 +196,9 @@ static int block_alloc_contiguous(struct sst_module *module,
 		size -= block->size;
 	}
 
+	list_for_each_entry(block, &tmp, list)
+		list_add(&block->module_list, &module->block_list);
+
 	list_splice(&tmp, &dsp->used_block_list);
 	return 0;
 }
@@ -247,8 +244,7 @@ static int block_alloc(struct sst_module *module,
 		/* do we span > 1 blocks */
 		if (data->size > block->size) {
 			ret = block_alloc_contiguous(module, data,
-				block->offset + block->size,
-				data->size - block->size);
+				block->offset, data->size);
 			if (ret == 0)
 				return ret;
 		}
@@ -344,7 +340,7 @@ static int block_alloc_fixed(struct sst_module *module,
 
 			err = block_alloc_contiguous(module, data,
 				block->offset + block->size,
-				data->size - block->size + data->offset - block->offset);
+				data->size - block->size);
 			if (err < 0)
 				return -ENOMEM;
 
@@ -371,15 +367,10 @@ static int block_alloc_fixed(struct sst_module *module,
 		if (data->offset >= block->offset && data->offset < block_end) {
 
 			err = block_alloc_contiguous(module, data,
-				block->offset + block->size,
-				data->size - block->size);
+				block->offset, data->size);
 			if (err < 0)
 				return -ENOMEM;
 
-			/* add block */
-			block->data_type = data->data_type;
-			list_move(&block->list, &dsp->used_block_list);
-			list_add(&block->module_list, &module->block_list);
 			return 0;
 		}
 
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index f5ebf36af889..535f517629fd 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -433,7 +433,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 	int ret = -ENODEV, i, j, region_count;
 	u32 offset, size;
 
-	dev = sst->dev;
+	dev = sst->dma_dev;
 
 	switch (sst->id) {
 	case SST_DEV_ID_LYNX_POINT:
@@ -466,7 +466,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 		return ret;
 	}
 
-	ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32));
+	ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(31));
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 50e4246d4b57..e7996b39a484 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -1159,11 +1159,14 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
 	void *data)
 {
 	struct sst_hsw_stream *stream;
+	struct sst_dsp *sst = hsw->dsp;
+	unsigned long flags;
 
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (stream == NULL)
 		return NULL;
 
+	spin_lock_irqsave(&sst->spinlock, flags);
 	list_add(&stream->node, &hsw->stream_list);
 	stream->notify_position = notify_position;
 	stream->pdata = data;
@@ -1172,6 +1175,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
 
 	/* work to process notification messages */
 	INIT_WORK(&stream->notify_work, hsw_notification_work);
+	spin_unlock_irqrestore(&sst->spinlock, flags);
 
 	return stream;
 }
@@ -1180,6 +1184,8 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 {
 	u32 header;
 	int ret = 0;
+	struct sst_dsp *sst = hsw->dsp;
+	unsigned long flags;
 
 	/* dont free DSP streams that are not commited */
 	if (!stream->commited)
@@ -1201,8 +1207,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 	trace_hsw_stream_free_req(stream, &stream->free_req);
 
 out:
+	cancel_work_sync(&stream->notify_work);
+	spin_lock_irqsave(&sst->spinlock, flags);
 	list_del(&stream->node);
 	kfree(stream);
+	spin_unlock_irqrestore(&sst->spinlock, flags);
 
 	return ret;
 }
@@ -1538,10 +1547,28 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 }
 
 /* Stream pointer positions */
-int sst_hsw_get_dsp_position(struct sst_hsw *hsw,
+u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream)
 {
-	return stream->rpos.position;
+	u32 rpos;
+
+	sst_dsp_read(hsw->dsp, &rpos,
+		stream->reply.read_position_register_address, sizeof(rpos));
+
+	return rpos;
+}
+
+/* Stream presentation (monotonic) positions */
+u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
+	struct sst_hsw_stream *stream)
+{
+	u64 ppos;
+
+	sst_dsp_read(hsw->dsp, &ppos,
+		stream->reply.presentation_position_register_address,
+		sizeof(ppos));
+
+	return ppos;
 }
 
 int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h
index d517929ccc38..2ac194a6d04b 100644
--- a/sound/soc/intel/sst-haswell-ipc.h
+++ b/sound/soc/intel/sst-haswell-ipc.h
@@ -464,7 +464,9 @@ int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 *position);
 int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 position);
-int sst_hsw_get_dsp_position(struct sst_hsw *hsw,
+u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw,
+	struct sst_hsw_stream *stream);
+u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream);
 
 /* HW port config */
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index 0a32dd13a23d..9d5f64a583a3 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -99,6 +99,7 @@ struct hsw_pcm_data {
 	struct snd_compr_stream *cstream;
 	unsigned int wpos;
 	struct mutex mutex;
+	bool allocated;
 };
 
 /* private data for the driver */
@@ -107,12 +108,14 @@ struct hsw_priv_data {
 	struct sst_hsw *hsw;
 
 	/* page tables */
-	unsigned char *pcm_pg[HSW_PCM_COUNT][2];
+	struct snd_dma_buffer dmab[HSW_PCM_COUNT][2];
 
 	/* DAI data */
 	struct hsw_pcm_data pcm[HSW_PCM_COUNT];
 };
 
+static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data);
+
 static inline u32 hsw_mixer_to_ipc(unsigned int value)
 {
 	if (value >= ARRAY_SIZE(volume_map))
@@ -273,28 +276,26 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = {
 };
 
 /* Create DMA buffer page table for DSP */
-static int create_adsp_page_table(struct hsw_priv_data *pdata,
-	struct snd_soc_pcm_runtime *rtd,
-	unsigned char *dma_area, size_t size, int pcm, int stream)
+static int create_adsp_page_table(struct snd_pcm_substream *substream,
+	struct hsw_priv_data *pdata, struct snd_soc_pcm_runtime *rtd,
+	unsigned char *dma_area, size_t size, int pcm)
 {
-	int i, pages;
+	struct snd_dma_buffer *dmab = snd_pcm_get_dma_buf(substream);
+	int i, pages, stream = substream->stream;
 
-	if (size % PAGE_SIZE)
-		pages = (size / PAGE_SIZE) + 1;
-	else
-		pages = size / PAGE_SIZE;
+	pages = snd_sgbuf_aligned_pages(size);
 
 	dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n",
 		dma_area, size, pages);
 
 	for (i = 0; i < pages; i++) {
 		u32 idx = (((i << 2) + i)) >> 1;
-		u32 pfn = (virt_to_phys(dma_area + i * PAGE_SIZE)) >> PAGE_SHIFT;
+		u32 pfn = snd_sgbuf_get_addr(dmab, i * PAGE_SIZE) >> PAGE_SHIFT;
 		u32 *pg_table;
 
 		dev_dbg(rtd->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn);
 
-		pg_table = (u32*)(pdata->pcm_pg[pcm][stream] + idx);
+		pg_table = (u32 *)(pdata->dmab[pcm][stream].area + idx);
 
 		if (i & 1)
 			*pg_table |= (pfn << 4);
@@ -317,12 +318,36 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct sst_hsw *hsw = pdata->hsw;
 	struct sst_module *module_data;
 	struct sst_dsp *dsp;
+	struct snd_dma_buffer *dmab;
 	enum sst_hsw_stream_type stream_type;
 	enum sst_hsw_stream_path_id path_id;
 	u32 rate, bits, map, pages, module_id;
 	u8 channels;
 	int ret;
 
+	/* check if we are being called a subsequent time */
+	if (pcm_data->allocated) {
+		ret = sst_hsw_stream_reset(hsw, pcm_data->stream);
+		if (ret < 0)
+			dev_dbg(rtd->dev, "error: reset stream failed %d\n",
+				ret);
+
+		ret = sst_hsw_stream_free(hsw, pcm_data->stream);
+		if (ret < 0) {
+			dev_dbg(rtd->dev, "error: free stream failed %d\n",
+				ret);
+			return ret;
+		}
+		pcm_data->allocated = false;
+
+		pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id,
+			hsw_notify_pointer, pcm_data);
+		if (pcm_data->stream == NULL) {
+			dev_err(rtd->dev, "error: failed to create stream\n");
+			return -EINVAL;
+		}
+	}
+
 	/* stream direction */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		path_id = SST_HSW_STREAM_PATH_SSP0_OUT;
@@ -416,8 +441,10 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	ret = create_adsp_page_table(pdata, rtd, runtime->dma_area,
-		runtime->dma_bytes, rtd->cpu_dai->id, substream->stream);
+	dmab = snd_pcm_get_dma_buf(substream);
+
+	ret = create_adsp_page_table(substream, pdata, rtd, runtime->dma_area,
+		runtime->dma_bytes, rtd->cpu_dai->id);
 	if (ret < 0)
 		return ret;
 
@@ -430,9 +457,9 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
 		pages = runtime->dma_bytes / PAGE_SIZE;
 
 	ret = sst_hsw_stream_buffer(hsw, pcm_data->stream,
-		virt_to_phys(pdata->pcm_pg[rtd->cpu_dai->id][substream->stream]),
+		pdata->dmab[rtd->cpu_dai->id][substream->stream].addr,
 		pages, runtime->dma_bytes, 0,
-		(u32)(virt_to_phys(runtime->dma_area) >> PAGE_SHIFT));
+		snd_sgbuf_get_addr(dmab, 0) >> PAGE_SHIFT);
 	if (ret < 0) {
 		dev_err(rtd->dev, "error: failed to set DMA buffer %d\n", ret);
 		return ret;
@@ -474,6 +501,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
 		dev_err(rtd->dev, "error: failed to commit stream %d\n", ret);
 		return ret;
 	}
+	pcm_data->allocated = true;
 
 	ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1);
 	if (ret < 0)
@@ -541,12 +569,14 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream)
 	struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
 	struct sst_hsw *hsw = pdata->hsw;
 	snd_pcm_uframes_t offset;
+	uint64_t ppos;
+	u32 position = sst_hsw_get_dsp_position(hsw, pcm_data->stream);
 
-	offset = bytes_to_frames(runtime,
-		sst_hsw_get_dsp_position(hsw, pcm_data->stream));
+	offset = bytes_to_frames(runtime, position);
+	ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream);
 
-	dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n",
-		frames_to_bytes(runtime, (u32)offset));
+	dev_dbg(rtd->dev, "PCM: DMA pointer %du bytes, pos %llu\n",
+		position, ppos);
 	return offset;
 }
 
@@ -606,6 +636,7 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream)
 		dev_dbg(rtd->dev, "error: free stream failed %d\n", ret);
 		goto out;
 	}
+	pcm_data->allocated = 0;
 	pcm_data->stream = NULL;
 
 out:
@@ -621,7 +652,7 @@ static struct snd_pcm_ops hsw_pcm_ops = {
 	.hw_free	= hsw_pcm_hw_free,
 	.trigger	= hsw_pcm_trigger,
 	.pointer	= hsw_pcm_pointer,
-	.mmap		= snd_pcm_lib_default_mmap,
+	.page		= snd_pcm_sgbuf_ops_page,
 };
 
 static void hsw_pcm_free(struct snd_pcm *pcm)
@@ -632,17 +663,16 @@ static void hsw_pcm_free(struct snd_pcm *pcm)
 static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_pcm *pcm = rtd->pcm;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct sst_pdata *pdata = dev_get_platdata(platform->dev);
+	struct device *dev = pdata->dma_dev;
 	int ret = 0;
 
-	ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		return ret;
-
 	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
 			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = snd_pcm_lib_preallocate_pages_for_all(pcm,
-			SNDRV_DMA_TYPE_DEV,
-			rtd->card->dev,
+			SNDRV_DMA_TYPE_DEV_SG,
+			dev,
 			hsw_pcm_hardware.buffer_bytes_max,
 			hsw_pcm_hardware.buffer_bytes_max);
 		if (ret) {
@@ -742,11 +772,14 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform)
 {
 	struct sst_pdata *pdata = dev_get_platdata(platform->dev);
 	struct hsw_priv_data *priv_data;
-	int i;
+	struct device *dma_dev;
+	int i, ret = 0;
 
 	if (!pdata)
 		return -ENODEV;
 
+	dma_dev = pdata->dma_dev;
+
 	priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);
 	priv_data->hsw = pdata->dsp;
 	snd_soc_platform_set_drvdata(platform, priv_data);
@@ -758,15 +791,17 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform)
 
 		/* playback */
 		if (hsw_dais[i].playback.channels_min) {
-			priv_data->pcm_pg[i][0] = kzalloc(PAGE_SIZE, GFP_DMA);
-			if (priv_data->pcm_pg[i][0] == NULL)
+			ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev,
+				PAGE_SIZE, &priv_data->dmab[i][0]);
+			if (ret < 0)
 				goto err;
 		}
 
 		/* capture */
 		if (hsw_dais[i].capture.channels_min) {
-			priv_data->pcm_pg[i][1] = kzalloc(PAGE_SIZE, GFP_DMA);
-			if (priv_data->pcm_pg[i][1] == NULL)
+			ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev,
+				PAGE_SIZE, &priv_data->dmab[i][1]);
+			if (ret < 0)
 				goto err;
 		}
 	}
@@ -776,11 +811,11 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform)
 err:
 	for (;i >= 0; i--) {
 		if (hsw_dais[i].playback.channels_min)
-			kfree(priv_data->pcm_pg[i][0]);
+			snd_dma_free_pages(&priv_data->dmab[i][0]);
 		if (hsw_dais[i].capture.channels_min)
-			kfree(priv_data->pcm_pg[i][1]);
+			snd_dma_free_pages(&priv_data->dmab[i][1]);
 	}
-	return -ENOMEM;
+	return ret;
 }
 
 static int hsw_pcm_remove(struct snd_soc_platform *platform)
@@ -791,9 +826,9 @@ static int hsw_pcm_remove(struct snd_soc_platform *platform)
 
 	for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) {
 		if (hsw_dais[i].playback.channels_min)
-			kfree(priv_data->pcm_pg[i][0]);
+			snd_dma_free_pages(&priv_data->dmab[i][0]);
 		if (hsw_dais[i].capture.channels_min)
-			kfree(priv_data->pcm_pg[i][1]);
+			snd_dma_free_pages(&priv_data->dmab[i][1]);
 	}
 
 	return 0;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 215b668166be..89424470a1f3 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -197,13 +197,12 @@ static void rsnd_dma_complete(void *data)
 	 * rsnd_dai_pointer_update() will be called twice,
 	 * ant it will breaks io->byte_pos
 	 */
-
-	rsnd_dai_pointer_update(io, io->byte_per_period);
-
 	if (dma->submit_loop)
 		rsnd_dma_continue(dma);
 
 	rsnd_unlock(priv, flags);
+
+	rsnd_dai_pointer_update(io, io->byte_per_period);
 }
 
 static void __rsnd_dma_start(struct rsnd_dma *dma)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7769b0a2bc5a..6d6ceee447d5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1612,8 +1612,11 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie)
 				"ASoC: Failed to turn on bias: %d\n", ret);
 	}
 
-	/* Prepare for a STADDBY->ON or ON->STANDBY transition */
-	if (d->bias_level != d->target_bias_level) {
+	/* Prepare for a transition to ON or away from ON */
+	if ((d->target_bias_level == SND_SOC_BIAS_ON &&
+	     d->bias_level != SND_SOC_BIAS_ON) ||
+	    (d->target_bias_level != SND_SOC_BIAS_ON &&
+	     d->bias_level == SND_SOC_BIAS_ON)) {
 		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
 		if (ret != 0)
 			dev_err(d->dev,
@@ -3475,8 +3478,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 		cpu_dai = rtd->cpu_dai;
 		codec_dai = rtd->codec_dai;
 
-		/* dynamic FE links have no fixed DAI mapping */
-		if (rtd->dai_link->dynamic)
+		/*
+		 * dynamic FE links have no fixed DAI mapping.
+		 * CODEC<->CODEC links have no direct connection.
+		 */
+		if (rtd->dai_link->dynamic || rtd->dai_link->params)
 			continue;
 
 		/* there is no point in connecting BE DAI links with dummies */
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2cedf09f6d96..a391de058037 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1675,7 +1675,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 			be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
 			break;
 		case SNDRV_PCM_TRIGGER_SUSPEND:
-			if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)
+			if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
 				continue;
 
 			if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))