Merge tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been major modernization with the standard bus: in ALSA
  sequencer core and HD-audio.  Also, HD-audio receives the regmap
  support replacing the in-house cache register cache code.  These
  changes shouldn't impact the existing behavior, but rather
  refactoring.

  In addition, HD-audio got the code split to a core library part and
  the "legacy" driver parts.  This is a preliminary work for adapting
  the upcoming ASoC HD-audio driver, and the whole transition is still
  work in progress, likely finished in 4.1.

  Along with them, there are many updates in ASoC area as usual, too:
  lots of cleanups, Intel code shuffling, etc.

  Here are some highlights:

  ALSA core:
   - PCM: the audio timestamp / wallclock enhancement
   - PCM: fixes in DPCM management
   - Fixes / cleanups of user-space control element management
   - Sequencer: modernization using the standard bus

  HD-audio:
   - Modernization using the standard bus
   - Regmap support
   - Use standard runtime PM for codec power saving
   - Widget-path based power-saving for IDT, VIA and Realtek codecs
   - Reorganized sysfs entries for each codec object
   - More Dell headset support

  ASoC:
   - Move of jack registration to the card level
   - Lots of ASoC cleanups, mainly moving things from the CODEC level to
     the card level
   - Support for DAPM routes specified by both the machine driver and DT
   - Continuing improvements to rcar
   - pcm512x enhacements
   - Intel platforms updates
   - rt5670 updates / fixes
   - New platforms / devices: some non-DSP Qualcomm platforms, Google's
     Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC

  Misc:
   - ice1724: Improved ESI W192M support
   - emu10k1: Emu 1010 fixes/enhancement"

* tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
  ALSA: hda - set GET bit when adding a vendor verb to the codec regmap
  ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450
  ALSA: hda - Fix another race in runtime PM refcounting
  ALSA: hda - Expose codec type sysfs
  ALSA: ctl: fix to handle several elements added by one operation for userspace element
  ASoC: Intel: fix array_size.cocci warnings
  ASoC: n810: Automatically disconnect non-connected pins
  ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
  ASoC: davinci-evm: Use card DAPM context to access widgets
  ASoC: mop500_ab8500: Use card DAPM context to access widgets
  ASoC: wm1133-ev1: Use card DAPM context to access widgets
  ASoC: atmel: Improve machine driver compile test coverage
  ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
  ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_*
  ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate
  ASoC: rnsd: fix build regression without CONFIG_OF
  ALSA: emu10k1: add toggles for E-mu 1010 optical ports
  ALSA: ctl: fill identical information to return value when adding userspace elements
  ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls
  ALSA: ctl: confirm to return all identical information in 'activate' event
  ...

8 files changed, 30 insertions, 39 deletions

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 081e406b3713..19c302b0d763 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -24,7 +24,7 @@ config SND_SOC_FSL_SAI
 	  in-tree drivers select it automatically.
 
 config SND_SOC_FSL_SSI
-	tristate "Synchronous Serial Interface module support"
+	tristate "Synchronous Serial Interface module (SSI) support"
 	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
 	select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
 	select REGMAP_MMIO
@@ -35,7 +35,7 @@ config SND_SOC_FSL_SSI
 	  in-tree drivers select it automatically.
 
 config SND_SOC_FSL_SPDIF
-	tristate "Sony/Philips Digital Interface module support"
+	tristate "Sony/Philips Digital Interface (S/PDIF) module support"
 	select REGMAP_MMIO
 	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
 	select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 3f6959c8e2f7..de438871040b 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -512,6 +512,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	memcpy(priv->dai_link, fsl_asoc_card_dai,
 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
 
+	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
+		goto asrc_fail;
+	}
+
 	/* Normal DAI Link */
 	priv->dai_link[0].cpu_of_node = cpu_np;
 	priv->dai_link[0].codec_of_node = codec_np;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 6b0c8f717ec2..e8bb8eef1d16 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1288,7 +1288,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	const struct of_device_id *of_id;
 	const char *p, *sprop;
 	const uint32_t *iprop;
-	struct resource res;
+	struct resource *res;
 	void __iomem *iomem;
 	char name[64];
 
@@ -1335,19 +1335,11 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 	ssi_private->cpu_dai_drv.name = dev_name(&pdev->dev);
 
-	/* Get the addresses and IRQ */
-	ret = of_address_to_resource(np, 0, &res);
-	if (ret) {
-		dev_err(&pdev->dev, "could not determine device resources\n");
-		return ret;
-	}
-	ssi_private->ssi_phys = res.start;
-
-	iomem = devm_ioremap(&pdev->dev, res.start, resource_size(&res));
-	if (!iomem) {
-		dev_err(&pdev->dev, "could not map device resources\n");
-		return -ENOMEM;
-	}
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	iomem = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(iomem))
+		return PTR_ERR(iomem);
+	ssi_private->ssi_phys = res->start;
 
 	ret = of_property_match_string(np, "clock-names", "ipg");
 	if (ret < 0) {
@@ -1393,8 +1385,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 			return ret;
 	}
 
-	ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component,
-					 &ssi_private->cpu_dai_drv, 1);
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component,
+					      &ssi_private->cpu_dai_drv, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
 		goto error_asoc_register;
@@ -1407,13 +1399,13 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		if (ret < 0) {
 			dev_err(&pdev->dev, "could not claim irq %u\n",
 					ssi_private->irq);
-			goto error_irq;
+			goto error_asoc_register;
 		}
 	}
 
 	ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev);
 	if (ret)
-		goto error_irq;
+		goto error_asoc_register;
 
 	/*
 	 * If codec-handle property is missing from SSI node, we assume
@@ -1454,9 +1446,6 @@ done:
 error_sound_card:
 	fsl_ssi_debugfs_remove(&ssi_private->dbg_stats);
 
-error_irq:
-	snd_soc_unregister_component(&pdev->dev);
-
 error_asoc_register:
 	if (ssi_private->soc->imx)
 		fsl_ssi_imx_clean(pdev, ssi_private);
@@ -1472,7 +1461,6 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 
 	if (ssi_private->pdev)
 		platform_device_unregister(ssi_private->pdev);
-	snd_soc_unregister_component(&pdev->dev);
 
 	if (ssi_private->soc->imx)
 		fsl_ssi_imx_clean(pdev, ssi_private);
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index f8cf10e16ce9..20e7400e2611 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -53,9 +53,9 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
 
 	/* Headphone jack detection */
 	if (gpio_is_valid(data->jack_gpio)) {
-		ret = snd_soc_jack_new(rtd->codec, "Headphone",
-				       SND_JACK_HEADPHONE | SND_JACK_BTN_0,
-				       &headset_jack);
+		ret = snd_soc_card_jack_new(rtd->card, "Headphone",
+					    SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+					    &headset_jack, NULL, 0);
 		if (ret)
 			return ret;
 
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 08d2a8069b0a..0bab76051fd8 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -326,7 +326,7 @@ static int psc_ac97_of_remove(struct platform_device *op)
 }
 
 /* Match table for of_platform binding */
-static struct of_device_id psc_ac97_match[] = {
+static const struct of_device_id psc_ac97_match[] = {
 	{ .compatible = "fsl,mpc5200-psc-ac97", },
 	{ .compatible = "fsl,mpc5200b-psc-ac97", },
 	{}
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 51fb0c00fe73..d8232943ccb6 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -217,7 +217,7 @@ static int psc_i2s_of_remove(struct platform_device *op)
 }
 
 /* Match table for of_platform binding */
-static struct of_device_id psc_i2s_match[] = {
+static const struct of_device_id psc_i2s_match[] = {
 	{ .compatible = "fsl,mpc5200-psc-i2s", },
 	{ .compatible = "fsl,mpc5200b-psc-i2s", },
 	{}
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index c44459d24c50..ec731223cab3 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -113,7 +113,7 @@ static int pcm030_fabric_remove(struct platform_device *op)
 	return ret;
 }
 
-static struct of_device_id pcm030_audio_match[] = {
+static const struct of_device_id pcm030_audio_match[] = {
 	{ .compatible = "phytec,pcm030-audio-fabric", },
 	{}
 };
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index a958937ab405..b454972dce35 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -202,23 +202,20 @@ static struct snd_soc_jack_pin mic_jack_pins[] = {
 static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Headphone jack detection */
-	snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
-	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
-			      hp_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
+			      &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
 	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
 
 	/* Microphone jack detection */
-	snd_soc_jack_new(codec, "Microphone",
-			 SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
-	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
-			      mic_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Microphone",
+			      SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
+			      mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
 	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
 			       SND_JACK_BTN_0);
 
-	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+	snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias");
 
 	return 0;
 }