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

…el/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.
  ...

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>

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, "???");

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);

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");

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),

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;
 	}

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);
 }

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 */

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;
 }
 

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];

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;
 

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;
 

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;
 }

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;

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;

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;

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;
 		}
 

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;