Merge tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "As the diffstat shows we've had again a lot of works done for this
  cycle: the majority of changes are the continued componentization and
  code refactoring in ASoC, the tree-wide PCM API updates and cleanups
  and SOF updates while a few ASoC driver updates are seen, too.

  Here we go, some highlights:

  Core:
   - Finally y2038 support landed to ALSA ABI; some ioctls have been
     extended and lots of tricks were applied
   - Applying the new managed PCM buffer API to all drivers; the API
     itself was already merged in 5.5
   - The already deprecated dimension support in ALSA control API is
     dropped completely now
   - Verification of ALSA control elements to catch API misuses

  ASoC:
   - Further code refactorings and moving things to the component level
   - Lots of updates and improvements on SOF / Intel drivers; now
     including common HDMI driver and SoundWire support
   - New driver support for Ingenic JZ4770, Mediatek MT6660, Qualcomm
     WCD934x and WSA881x, and Realtek RT700, RT711, RT715, RT1011,
     RT1015 and RT1308

  HD-audio:
   - Improved ring-buffer communications using waitqueue
   - Drop the superfluous buffer preallocation on x86

  Others:
   - Many code cleanups, mostly constifications over the whole tree
   - USB-audio: quirks for MOTU, Corsair Virtuoso, Line6 Helix
   - FireWire: code refactoring for oxfw and dice drivers"

* tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (638 commits)
  ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
  ALSA: hda: Add Clevo W65_67SB the power_save blacklist
  ASoC: soc-core: remove null_snd_soc_ops
  ASoC: soc-pcm: add soc_rtd_trigger()
  ASoC: soc-pcm: add soc_rtd_hw_free()
  ASoC: soc-pcm: add soc_rtd_hw_params()
  ASoC: soc-pcm: add soc_rtd_prepare()
  ASoC: soc-pcm: add soc_rtd_shutdown()
  ASoC: soc-pcm: add soc_rtd_startup()
  ASoC: rt1015: add rt1015 amplifier driver
  ASoC: madera: Correct some kernel doc
  ASoC: topology: fix soc_tplg_fe_link_create() - link->dobj initialization order
  ASoC: Intel: skl_hda_dsp_common: Fix global-out-of-bounds bug
  ASoC: madera: Correct DMIC only input hook ups
  ALSA: cs46xx: fix spelling mistake "to" -> "too"
  ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
  ASoC: Add MediaTek MT6660 Speaker Amp Driver
  ASoC: dt-bindings: rt5645: add suppliers
  ASoC: max98090: fix deadlock in max98090_dapm_put_enum_double()
  ASoC: dapm: add snd_soc_dapm_put_enum_double_locked
  ...

4 files changed, 46 insertions, 21 deletions

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 8e5371801d88..e1e937eb1dc1 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -664,18 +664,39 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
 	pm_runtime_get_sync(mcasp->dev);
-	if (dir == SND_SOC_CLOCK_OUT) {
+
+	if (dir == SND_SOC_CLOCK_IN) {
+		switch (clk_id) {
+		case MCASP_CLK_HCLK_AHCLK:
+			mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
+				       AHCLKXE);
+			mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
+				       AHCLKRE);
+			clear_bit(PIN_BIT_AHCLKX, &mcasp->pdir);
+			break;
+		case MCASP_CLK_HCLK_AUXCLK:
+			mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
+				       AHCLKXE);
+			mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
+				       AHCLKRE);
+			set_bit(PIN_BIT_AHCLKX, &mcasp->pdir);
+			break;
+		default:
+			dev_err(mcasp->dev, "Invalid clk id: %d\n", clk_id);
+			goto out;
+		}
+	} else {
+		/* Select AUXCLK as HCLK */
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
 		set_bit(PIN_BIT_AHCLKX, &mcasp->pdir);
-	} else {
-		mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
-		mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
-		clear_bit(PIN_BIT_AHCLKX, &mcasp->pdir);
 	}
-
+	/*
+	 * When AHCLK X/R is selected to be output it means that the HCLK is
+	 * the same clock - coming via AUXCLK.
+	 */
 	mcasp->sysclk_freq = freq;
-
+out:
 	pm_runtime_put(mcasp->dev);
 	return 0;
 }
diff --git a/sound/soc/ti/davinci-mcasp.h b/sound/soc/ti/davinci-mcasp.h
index bc705d6ca48b..5de2b8a31061 100644
--- a/sound/soc/ti/davinci-mcasp.h
+++ b/sound/soc/ti/davinci-mcasp.h
@@ -295,6 +295,10 @@
 #define NUMEVT(x)	(((x) & 0xFF) << 8)
 #define NUMDMA_MASK	(0xFF)
 
+/* Source of High-frequency transmit/receive clock */
+#define MCASP_CLK_HCLK_AHCLK		0 /* AHCLKX/R */
+#define MCASP_CLK_HCLK_AUXCLK		1 /* Internal functional clock */
+
 /* clock divider IDs */
 #define MCASP_CLKDIV_AUXCLK		0 /* HCLK divider from AUXCLK */
 #define MCASP_CLKDIV_BCLK		1 /* BCLK divider from HCLK */
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index b8c8290265c7..a726cd7a8252 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -458,11 +458,11 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
 }
 
 #ifdef CONFIG_PM_SLEEP
-static int omap_mcpdm_suspend(struct snd_soc_dai *dai)
+static int omap_mcpdm_suspend(struct snd_soc_component *component)
 {
-	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	struct omap_mcpdm *mcpdm = snd_soc_component_get_drvdata(component);
 
-	if (dai->active) {
+	if (component->active) {
 		omap_mcpdm_stop(mcpdm);
 		omap_mcpdm_close_streams(mcpdm);
 	}
@@ -476,15 +476,15 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static int omap_mcpdm_resume(struct snd_soc_dai *dai)
+static int omap_mcpdm_resume(struct snd_soc_component *component)
 {
-	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	struct omap_mcpdm *mcpdm = snd_soc_component_get_drvdata(component);
 
 	if (mcpdm->pm_active_count) {
 		while (mcpdm->pm_active_count--)
 			pm_runtime_get_sync(mcpdm->dev);
 
-		if (dai->active) {
+		if (component->active) {
 			omap_mcpdm_open_streams(mcpdm);
 			omap_mcpdm_start(mcpdm);
 		}
@@ -504,8 +504,6 @@ static int omap_mcpdm_resume(struct snd_soc_dai *dai)
 static struct snd_soc_dai_driver omap_mcpdm_dai = {
 	.probe = omap_mcpdm_probe,
 	.remove = omap_mcpdm_remove,
-	.suspend = omap_mcpdm_suspend,
-	.resume = omap_mcpdm_resume,
 	.probe_order = SND_SOC_COMP_ORDER_LATE,
 	.remove_order = SND_SOC_COMP_ORDER_EARLY,
 	.playback = {
@@ -527,6 +525,8 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = {
 
 static const struct snd_soc_component_driver omap_mcpdm_component = {
 	.name		= "omap-mcpdm",
+	.suspend	= omap_mcpdm_suspend,
+	.resume		= omap_mcpdm_resume,
 };
 
 void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 588f680a9c24..fdb0dc85fe67 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -328,11 +328,11 @@ static struct snd_soc_aux_dev rx51_aux_dev[] = {
 
 static struct snd_soc_codec_conf rx51_codec_conf[] = {
 	{
-		.dev_name = "tlv320aic3x-codec.2-0019",
+		.dlc = COMP_CODEC_CONF("tlv320aic3x-codec.2-0019"),
 		.name_prefix = "b",
 	},
 	{
-		.dev_name = "tpa6130a2.2-0060",
+		.dlc = COMP_CODEC_CONF("tpa6130a2.2-0060"),
 		.name_prefix = "TPA6130A2",
 	},
 };
@@ -397,8 +397,8 @@ static int rx51_soc_probe(struct platform_device *pdev)
 		}
 		rx51_aux_dev[0].dlc.name = NULL;
 		rx51_aux_dev[0].dlc.of_node = dai_node;
-		rx51_codec_conf[0].dev_name = NULL;
-		rx51_codec_conf[0].of_node = dai_node;
+		rx51_codec_conf[0].dlc.name = NULL;
+		rx51_codec_conf[0].dlc.of_node = dai_node;
 
 		dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0);
 		if (!dai_node) {
@@ -407,8 +407,8 @@ static int rx51_soc_probe(struct platform_device *pdev)
 		}
 		rx51_aux_dev[1].dlc.name = NULL;
 		rx51_aux_dev[1].dlc.of_node = dai_node;
-		rx51_codec_conf[1].dev_name = NULL;
-		rx51_codec_conf[1].of_node = dai_node;
+		rx51_codec_conf[1].dlc.name = NULL;
+		rx51_codec_conf[1].dlc.of_node = dai_node;
 	}
 
 	pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);