Merge tag 'sound-3.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "In this batch, you can find lots of cleanups through the whole
  subsystem, as our good New Year's resolution.  Lots of LOCs and
  commits are about LINE6 driver that was promoted finally from staging
  tree, and as usual, there've been widely spread ASoC changes.

  Here some highlights:

  ALSA core changes
   - Embedding struct device into ALSA core structures
   - sequencer core cleanups / fixes
   - PCM msbits constraints cleanups / fixes
   - New SNDRV_PCM_TRIGGER_DRAIN command
   - PCM kerneldoc fixes, header cleanups
   - PCM code cleanups using more standard codes
   - Control notification ID fixes

  Driver cleanups
   - Cleanups of PCI PM callbacks
   - Timer helper usages cleanups
   - Simplification (e.g. argument reduction) of many driver codes

  HD-audio
   - Hotkey and LED support on HP laptops with Realtek codecs
   - Dock station support on HP laptops
   - Toshiba Satellite S50D fixup
   - Enhanced wallclock timestamp handling for HD-audio
   - Componentization to simplify the linkage between i915 and hd-audio
     drivers for Intel HDMI/DP

  USB-audio
   - Akai MPC Element support
   - Enhanced timestamp handling

  ASoC
   - Lots of refactoringin ASoC core, moving drivers to more data driven
     initialization and rationalizing a lot of DAPM usage
   - Much improved handling of CDCLK clocks on Samsung I2S controllers
   - Lots of driver specific cleanups and feature improvements
   - CODEC support for TI PCM514x and TLV320AIC3104 devices
   - Board support for Tegra systems with Realtek RT5677
   - New driver for Maxim max98357a
   - More enhancements / fixes for Intel SST driver

  Others
   - Promotion of LINE6 driver from staging along with lots of rewrites
     and cleanups
   - DT support for old non-ASoC atmel driver
   - oxygen cleanups, XIO2001 init, Studio Evolution SE6x support
   - Emu8000 DRAM size detection fix on ISA(!!) AWE64 boards
   - A few more ak411x fixes for ice1724 boards"

* tag 'sound-3.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (542 commits)
  ALSA: line6: toneport: Use explicit type for firmware version
  ALSA: line6: Use explicit type for serial number
  ALSA: line6: Return EIO if read/write not successful
  ALSA: line6: Return error if device not responding
  ALSA: line6: Add delay before reading status
  ASoC: Intel: Clean data after SST fw fetch
  ALSA: hda - Add docking station support for another HP machine
  ALSA: control: fix failure to return new numerical ID in 'replace' event data
  ALSA: usb: update trigger timestamp on first non-zero URB submitted
  ALSA: hda: read trigger_timestamp immediately after starting DMA
  ALSA: pcm: allow for trigger_tstamp snapshot in .trigger
  ALSA: pcm: don't override timestamp unconditionally
  ALSA: off by one bug in snd_riptide_joystick_probe()
  ASoC: rt5670: Set use_single_rw flag for regmap
  ASoC: rt286: Add rt288 codec support
  ASoC: max98357a: Fix build in !CONFIG_OF case
  ASoC: Intel: fix platform_no_drv_owner.cocci warnings
  ARM: dts: Switch Odroid X2/U2 to simple-audio-card
  ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update
  ALSA: control: fix failure to return numerical ID in 'add' event
  ...

14 files changed, 27 insertions, 104 deletions

diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 9ce70fc67b09..e1aa3834b101 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -42,25 +42,6 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret;
 
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	/* fsl_ssi lacks the set_fmt ops. */
-	if (ret && ret != -ENOTSUPP) {
-		dev_err(cpu_dai->dev,
-			"Failed to set the cpu dai format.\n");
-		return ret;
-	}
-
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	if (ret) {
-		dev_err(cpu_dai->dev,
-			"Failed to set the codec format.\n");
-		return ret;
-	}
-
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
 				     CODEC_CLOCK, SND_SOC_CLOCK_OUT);
 	if (ret) {
@@ -69,7 +50,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
+	snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
 
 	ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
 				SND_SOC_CLOCK_IN);
@@ -91,6 +72,8 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
 	.codec_dai_name	= "tlv320aic23-hifi",
+	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &eukrea_tlv320_snd_ops,
 };
 
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 026a80117540..c068494bae30 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -818,7 +818,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	asrc_priv->pdev = pdev;
-	strncpy(asrc_priv->name, np->name, sizeof(asrc_priv->name) - 1);
 
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -837,12 +836,12 @@ static int fsl_asrc_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
 		return irq;
 	}
 
 	ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0,
-			       asrc_priv->name, asrc_priv);
+			       dev_name(&pdev->dev), asrc_priv);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret);
 		return ret;
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
index a3f211f53c23..4aed63c4b431 100644
--- a/sound/soc/fsl/fsl_asrc.h
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -433,7 +433,6 @@ struct fsl_asrc_pair {
  * @channel_avail: non-occupied channel numbers
  * @asrc_rate: default sample rate for ASoC Back-Ends
  * @asrc_width: default sample width for ASoC Back-Ends
- * @name: driver name
  */
 struct fsl_asrc {
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
@@ -452,8 +451,6 @@ struct fsl_asrc {
 
 	int asrc_rate;
 	int asrc_width;
-
-	char name[32];
 };
 
 extern struct snd_soc_platform_driver fsl_asrc_platform;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 1c08ab13637c..5c7597191e3f 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -774,7 +774,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
 		return irq;
 	}
 
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 032d2d33619c..ec79c3d5e65e 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -612,7 +612,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
 		return irq;
 	}
 
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index af0429421fc8..75870c0ea2c9 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -90,7 +90,6 @@ struct spdif_mixer_control {
  * @sysclk: system clock for rx clock rate measurement
  * @dma_params_tx: DMA parameters for transmit channel
  * @dma_params_rx: DMA parameters for receive channel
- * @name: driver name
  */
 struct fsl_spdif_priv {
 	struct spdif_mixer_control fsl_spdif_control;
@@ -109,12 +108,8 @@ struct fsl_spdif_priv {
 	struct clk *sysclk;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
-
-	/* The name space will be allocated dynamically */
-	char name[0];
 };
 
-
 /* DPLL locked and lock loss interrupt handler */
 static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
 {
@@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	if (!np)
 		return -ENODEV;
 
-	spdif_priv = devm_kzalloc(&pdev->dev,
-			sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1,
-			GFP_KERNEL);
+	spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL);
 	if (!spdif_priv)
 		return -ENOMEM;
 
-	strcpy(spdif_priv->name, np->name);
-
 	spdif_priv->pdev = pdev;
 
 	/* Initialize this copy of the CPU DAI driver structure */
 	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
-	spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+	spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev);
 
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -1198,12 +1189,12 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
 		return irq;
 	}
 
 	ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
-			spdif_priv->name, spdif_priv);
+			       dev_name(&pdev->dev), spdif_priv);
 	if (ret) {
 		dev_err(&pdev->dev, "could not claim irq %u\n", irq);
 		return ret;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 059496ed9ad7..2595611e8a6d 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -160,7 +160,7 @@ struct fsl_ssi_soc_data {
  */
 struct fsl_ssi_private {
 	struct regmap *regs;
-	unsigned int irq;
+	int irq;
 	struct snd_soc_dai_driver cpu_dai_drv;
 
 	unsigned int dai_fmt;
@@ -992,8 +992,8 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask,
 	regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN,
 			CCSR_SSI_SCR_SSIEN);
 
-	regmap_write(regs, CCSR_SSI_STMSK, tx_mask);
-	regmap_write(regs, CCSR_SSI_SRMSK, rx_mask);
+	regmap_write(regs, CCSR_SSI_STMSK, ~tx_mask);
+	regmap_write(regs, CCSR_SSI_SRMSK, ~rx_mask);
 
 	regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val);
 
@@ -1362,8 +1362,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi_private->irq = platform_get_irq(pdev, 0);
-	if (ssi_private->irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+	if (!ssi_private->irq) {
+		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
 		return ssi_private->irq;
 	}
 
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 2ac7755da876..b9e42b503a37 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -86,33 +86,6 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
 }
 EXPORT_SYMBOL(fsl_asoc_get_dma_channel);
 
-/**
- * fsl_asoc_xlate_tdm_slot_mask - generate TDM slot TX/RX mask.
- *
- * @slots: Number of slots in use.
- * @tx_mask: bitmask representing active TX slots.
- * @rx_mask: bitmask representing active RX slots.
- *
- * This function used to generate the TDM slot TX/RX mask. And the TX/RX
- * mask will use a 0 bit for an active slot as default, and the default
- * active bits are at the LSB of the mask value.
- */
-int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots,
-				    unsigned int *tx_mask,
-				    unsigned int *rx_mask)
-{
-	if (!slots)
-		return -EINVAL;
-
-	if (tx_mask)
-		*tx_mask = ~((1 << slots) - 1);
-	if (rx_mask)
-		*rx_mask = ~((1 << slots) - 1);
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(fsl_asoc_xlate_tdm_slot_mask);
-
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale ASoC utility code");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
index df535db40313..1687b66ef18e 100644
--- a/sound/soc/fsl/fsl_utils.h
+++ b/sound/soc/fsl/fsl_utils.h
@@ -22,7 +22,4 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name,
 			     struct snd_soc_dai_link *dai,
 			     unsigned int *dma_channel_id,
 			     unsigned int *dma_id);
-int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots,
-				    unsigned int *tx_mask,
-				    unsigned int *rx_mask);
 #endif /* _FSL_UTILS_H */
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 6bf5bce01a92..9e6493d4e7ff 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -37,8 +37,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	int ret;
 
-	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xfffffffc, 0xfffffffc,
-					4, 16);
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16);
 	if (ret)
 		return ret;
 
@@ -46,7 +45,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 2, 16);
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index e94704f1b9ee..33da26a12457 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
 	data->card.dev = &pdev->dev;
 	data->card.dai_link = &data->dai;
 	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
 
 	ret = snd_soc_of_parse_card_name(&data->card, "model");
 	if (ret)
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index fa801e17c51e..461ce27b884f 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -74,8 +74,8 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
 	sccr |= SSI_STCCR_DC(slots - 1);
 	writel(sccr, ssi->base + SSI_SRCCR);
 
-	writel(tx_mask, ssi->base + SSI_STMSK);
-	writel(rx_mask, ssi->base + SSI_SRMSK);
+	writel(~tx_mask, ssi->base + SSI_STMSK);
+	writel(~rx_mask, ssi->base + SSI_SRMSK);
 
 	return 0;
 }
@@ -340,7 +340,6 @@ static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = {
 	.set_fmt	= imx_ssi_set_dai_fmt,
 	.set_clkdiv	= imx_ssi_set_dai_clkdiv,
 	.set_sysclk	= imx_ssi_set_dai_sysclk,
-	.xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask,
 	.set_tdm_slot	= imx_ssi_set_dai_tdm_slot,
 	.trigger	= imx_ssi_trigger,
 };
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index b1ced7b8d80c..198eeb3f3f7a 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -55,16 +55,6 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret;
-	u32 dai_format;
-
-	dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
-		SND_SOC_DAIFMT_CBM_CFM;
-
-	/* set codec DAI configuration */
-	snd_soc_dai_set_fmt(codec_dai, dai_format);
-
-	/* set cpu DAI configuration */
-	snd_soc_dai_set_fmt(cpu_dai, dai_format);
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
 				     25000000, SND_SOC_CLOCK_OUT);
@@ -164,6 +154,8 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
 	.platform_name	= "imx-ssi.0",
 	.codec_name	= "tlv320aic32x4.0-0018",
 	.cpu_dai_name	= "imx-ssi.0",
+	.dai_fmt	= SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &mx27vis_aic32x4_snd_ops,
 };
 
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index 804749a6c61e..a958937ab405 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -87,7 +87,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
 	snd_pcm_format_t format = params_format(params);
 	unsigned int rate = params_rate(params);
 	unsigned int channels = params_channels(params);
-	u32 dai_format;
 
 	/* find the correct audio parameters */
 	for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
@@ -104,22 +103,13 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
 	/* codec FLL input is 14.75 MHz from MCLK */
 	snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
 
-	dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-		SND_SOC_DAIFMT_CBM_CFM;
-
-	/* set codec DAI configuration */
-	snd_soc_dai_set_fmt(codec_dai, dai_format);
-
-	/* set cpu DAI configuration */
-	snd_soc_dai_set_fmt(cpu_dai, dai_format);
-
 	/* TODO: The SSI driver should figure this out for us */
 	switch (channels) {
 	case 2:
-		snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
+		snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
 		break;
 	case 1:
-		snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
+		snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 1, 0);
 		break;
 	default:
 		return -EINVAL;
@@ -244,6 +234,8 @@ static struct snd_soc_dai_link wm1133_ev1_dai = {
 	.init = wm1133_ev1_init,
 	.ops = &wm1133_ev1_ops,
 	.symmetric_rates = 1,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
 };
 
 static struct snd_soc_card wm1133_ev1 = {