Merge tag 'sound-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been lots of changes in ALSA core, HD-audio and ASoC, also
  most of PCI drivers touched by conversions of printks.  All these
  resulted in a high volume and wide ranged patch sets in this release.
  Many changes are fairly trivial, but also lots of nice cleanups and
  refactors.  There are a few new drivers, most notably, the Intel
  Haswell and Baytrail ASoC driver.

  Core changes:
   - A bit modernization; embed the device struct into snd_card struct,
     so that it may be referred from the beginning.  A new
     snd_card_new() function is introduced for that, and all drivers
     have been converted.

   - Simplification in the device management code in ALSA core; now
     managed by a simple priority list instead

   - Converted many kernel messages to use the standard dev_err() & co;
     this would be the pretty visible difference, especially for
     HD-audio.

  HD-audio:
   - Conexant codecs use the auto-parser as default now; the old static
     code still remains in case of regressions.  Some old quirks have
     been rewritten with the fixups for auto-parser.

   - C-Media codecs also use the auto-parser as default now, too.

   - A device struct is assigned to each HD-audio codec, and the
     formerly hwdep attributes are accessible over the codec sysfs, too.
     hwdep attributes still remain for compatibility.

   - Split the PCI-specific stuff for HD-audio controller into a
     separate module, ane make a helper module for the generic
     controller driver.  This is a preliminary change for supporting
     Tegra HDMI controller in near future, which slipped from 3.15
     merge.

   - Device-specific fixes: mute LED support for Lenovo Ideapad, mic LED
     fix for HP laptops, more ASUS subwoofer quirks, yet more Dell
     laptop headset quirks

   - Make the HD-audio codec response a bit more robust

   - A few improvements on Realtek ALC282 / 283 about the pop noises

   - A couple of Intel HDMI fixes

  ASoC:
   - Lots of cleanups for enumerations; refactored lots of error prone
     original codes to use more modern APIs

   - Elimination of the ASoC level wrappers for I2C and SPI moving us
     closer to converting to regmap completely and avoiding some
     randconfig hassle

   - Provide both manually and transparently locked DAPM APIs rather
     than a mix of the two fixing some concurrency issues

   - Start converting CODEC drivers to use separate bus interface
     drivers rather than having them all in one file helping avoid
     dependency issues

   - DPCM support for Intel Haswell and Bay Trail platforms, lots of
     fixes

   - Lots of work on improvements for simple-card, DaVinci and the
     Renesas rcar drivers.

   - New drivers for Analog Devices ADAU1977, TI PCM512x and parts of
     the CSR SiRF SoC, TLV320AIC31XXX, Armada 370 DB, Cirrus cs42xx8

   - Fixes for the simple-card DAI format DT mess

   - DT support for a couple more devices.

   - Use of the tdm_slot mapping in a few drivers

  Others:
   - Support of reset_resume callback for improved S4 in USB-audio
     driver; the device with boot quirks have been little tested, which
     we need to watch out in this development cycle

   - Add PM support for ICE1712 driver (finally!); it's still pretty
     partial support, only for M-Audio devices"

* tag 'sound-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (610 commits)
  ALSA: ice1712: Add suspend support for M-Audio ICE1712-based cards
  ALSA: ice1712: add suspend support for ICE1712 chip
  ALSA: hda - Enable beep for ASUS 1015E
  ALSA: asihpi: fix some indenting in snd_card_asihpi_pcm_new()
  ALSA: hda - add headset mic detect quirks for three Dell laptops
  ASoC: tegra: move AC97 clock handling to the machine driver
  ASoC: simple-card: Handle many DAI links
  ASoC: simple-card: Add DT documentation for multi-DAI links
  ASoC: simple-card: dynamically allocate the DAI link and properties
  ASoC: imx-ssi: Add .xlate_tdm_slot_mask() support.
  ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support.
  ASoC: fsl-utils: Add fsl_asoc_xlate_tdm_slot_mask() support.
  ASoC: core: remove the 'of_' prefix of of_xlate_tdm_slot_mask.
  ASoC: rcar: subnode tidyup for renesas,rsnd.txt
  ASoC: Remove name_prefix unset during DAI link init hack
  ALSA: hda - Inform the unexpectedly ignored pins by auto-parser
  ASoC: rcar: bugfix: it cares about the non-src case
  ARM: bockw: fixup SND_SOC_DAIFMT_CBx_CFx flags
  ASoC: pcm: Drop incorrect double/extra frees
  ASoC: mfld_machine: Fix compile error
  ...

1 files changed, 52 insertions, 35 deletions

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 97db3b45b411..5522d2566c67 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1479,7 +1479,9 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 
 static int wm8962_dsp2_write_config(struct snd_soc_codec *codec)
 {
-	return regcache_sync_region(codec->control_data,
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+	return regcache_sync_region(wm8962->regmap,
 				    WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER);
 }
 
@@ -1658,16 +1660,16 @@ static const char *cap_hpf_mode_text[] = {
 	"Hi-fi", "Application"
 };
 
-static const struct soc_enum cap_hpf_mode =
-	SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(cap_hpf_mode,
+			    WM8962_ADC_DAC_CONTROL_2, 10, cap_hpf_mode_text);
 
 
 static const char *cap_lhpf_mode_text[] = {
 	"LPF", "HPF"
 };
 
-static const struct soc_enum cap_lhpf_mode =
-	SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(cap_lhpf_mode,
+			    WM8962_LHPF1, 1, cap_lhpf_mode_text);
 
 static const struct snd_kcontrol_new wm8962_snd_controls[] = {
 SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1),
@@ -2014,40 +2016,40 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
 
 static const char *st_text[] = { "None", "Left", "Right" };
 
-static const struct soc_enum str_enum =
-	SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
+static SOC_ENUM_SINGLE_DECL(str_enum,
+			    WM8962_DAC_DSP_MIXING_1, 2, st_text);
 
 static const struct snd_kcontrol_new str_mux =
 	SOC_DAPM_ENUM("Right Sidetone", str_enum);
 
-static const struct soc_enum stl_enum =
-	SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_2, 2, 3, st_text);
+static SOC_ENUM_SINGLE_DECL(stl_enum,
+			    WM8962_DAC_DSP_MIXING_2, 2, st_text);
 
 static const struct snd_kcontrol_new stl_mux =
 	SOC_DAPM_ENUM("Left Sidetone", stl_enum);
 
 static const char *outmux_text[] = { "DAC", "Mixer" };
 
-static const struct soc_enum spkoutr_enum =
-	SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_2, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(spkoutr_enum,
+			    WM8962_SPEAKER_MIXER_2, 7, outmux_text);
 
 static const struct snd_kcontrol_new spkoutr_mux =
 	SOC_DAPM_ENUM("SPKOUTR Mux", spkoutr_enum);
 
-static const struct soc_enum spkoutl_enum =
-	SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_1, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(spkoutl_enum,
+			    WM8962_SPEAKER_MIXER_1, 7, outmux_text);
 
 static const struct snd_kcontrol_new spkoutl_mux =
 	SOC_DAPM_ENUM("SPKOUTL Mux", spkoutl_enum);
 
-static const struct soc_enum hpoutr_enum =
-	SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_2, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(hpoutr_enum,
+			    WM8962_HEADPHONE_MIXER_2, 7, outmux_text);
 
 static const struct snd_kcontrol_new hpoutr_mux =
 	SOC_DAPM_ENUM("HPOUTR Mux", hpoutr_enum);
 
-static const struct soc_enum hpoutl_enum =
-	SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_1, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(hpoutl_enum,
+			    WM8962_HEADPHONE_MIXER_1, 7, outmux_text);
 
 static const struct snd_kcontrol_new hpoutl_mux =
 	SOC_DAPM_ENUM("HPOUTL Mux", hpoutl_enum);
@@ -2884,9 +2886,13 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
 	snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
 
-	try_wait_for_completion(&wm8962->fll_lock);
+	reinit_completion(&wm8962->fll_lock);
 
-	pm_runtime_get_sync(codec->dev);
+	ret = pm_runtime_get_sync(codec->dev);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to resume device: %d\n", ret);
+		return ret;
+	}
 
 	snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 			    WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
@@ -2894,8 +2900,6 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 
 	dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
 
-	ret = 0;
-
 	/* This should be a massive overestimate but go even
 	 * higher if we'll error out
 	 */
@@ -2909,14 +2913,17 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 
 	if (timeout == 0 && wm8962->irq) {
 		dev_err(codec->dev, "FLL lock timed out");
-		ret = -ETIMEDOUT;
+		snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
+				    WM8962_FLL_ENA, 0);
+		pm_runtime_put(codec->dev);
+		return -ETIMEDOUT;
 	}
 
 	wm8962->fll_fref = Fref;
 	wm8962->fll_fout = Fout;
 	wm8962->fll_src = source;
 
-	return ret;
+	return 0;
 }
 
 static int wm8962_mute(struct snd_soc_dai *dai, int mute)
@@ -3003,9 +3010,16 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 	unsigned int active;
 	int reg, ret;
 
+	ret = pm_runtime_get_sync(dev);
+	if (ret < 0) {
+		dev_err(dev, "Failed to resume: %d\n", ret);
+		return IRQ_NONE;
+	}
+
 	ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK,
 			  &mask);
 	if (ret != 0) {
+		pm_runtime_put(dev);
 		dev_err(dev, "Failed to read interrupt mask: %d\n",
 			ret);
 		return IRQ_NONE;
@@ -3013,14 +3027,17 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 
 	ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, &active);
 	if (ret != 0) {
+		pm_runtime_put(dev);
 		dev_err(dev, "Failed to read interrupt: %d\n", ret);
 		return IRQ_NONE;
 	}
 
 	active &= ~mask;
 
-	if (!active)
+	if (!active) {
+		pm_runtime_put(dev);
 		return IRQ_NONE;
+	}
 
 	/* Acknowledge the interrupts */
 	ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active);
@@ -3070,6 +3087,8 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 				   msecs_to_jiffies(250));
 	}
 
+	pm_runtime_put(dev);
+
 	return IRQ_HANDLED;
 }
 
@@ -3089,6 +3108,7 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int irq_mask, enable;
 
 	wm8962->jack = jack;
@@ -3109,14 +3129,18 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 	snd_soc_jack_report(wm8962->jack, 0,
 			    SND_JACK_MICROPHONE | SND_JACK_BTN_0);
 
+	snd_soc_dapm_mutex_lock(dapm);
+
 	if (jack) {
-		snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK");
-		snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS");
+		snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK");
+		snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
 	} else {
-		snd_soc_dapm_disable_pin(&codec->dapm, "SYSCLK");
-		snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "SYSCLK");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
 	}
 
+	snd_soc_dapm_mutex_unlock(dapm);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8962_mic_detect);
@@ -3400,13 +3424,6 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	bool dmicclk, dmicdat;
 
 	wm8962->codec = codec;
-	codec->control_data = wm8962->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	wm8962->disable_nb[0].notifier_call = wm8962_regulator_event_0;
 	wm8962->disable_nb[1].notifier_call = wm8962_regulator_event_1;