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, 29 insertions, 38 deletions

diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 1d9c2ed223bc..4a956d1cb269 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -44,48 +44,46 @@
 static int tosa_jack_func;
 static int tosa_spk_func;
 
-static void tosa_ext_control(struct snd_soc_codec *codec)
+static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_mutex_lock(dapm);
 
 	/* set up jack connection */
 	switch (tosa_jack_func) {
 	case TOSA_HP:
-		snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
-		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
-		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
 		break;
 	case TOSA_MIC_INT:
-		snd_soc_dapm_enable_pin(dapm, "Mic (Internal)");
-		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
-		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
 		break;
 	case TOSA_HEADSET:
-		snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
-		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
-		snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
 		break;
 	}
 
 	if (tosa_spk_func == TOSA_SPK_ON)
-		snd_soc_dapm_enable_pin(dapm, "Speaker");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
 	else
-		snd_soc_dapm_disable_pin(dapm, "Speaker");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+
+	snd_soc_dapm_sync_unlocked(dapm);
 
-	snd_soc_dapm_sync(dapm);
+	snd_soc_dapm_mutex_unlock(dapm);
 }
 
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
-
-	mutex_lock(&codec->mutex);
 
 	/* check the jack status at stream startup */
-	tosa_ext_control(codec);
-
-	mutex_unlock(&codec->mutex);
+	tosa_ext_control(&rtd->card->dapm);
 
 	return 0;
 }
@@ -104,13 +102,13 @@ static int tosa_get_jack(struct snd_kcontrol *kcontrol,
 static int tosa_set_jack(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 
 	if (tosa_jack_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	tosa_jack_func = ucontrol->value.integer.value[0];
-	tosa_ext_control(codec);
+	tosa_ext_control(&card->dapm);
 	return 1;
 }
 
@@ -124,13 +122,13 @@ static int tosa_get_spk(struct snd_kcontrol *kcontrol,
 static int tosa_set_spk(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 
 	if (tosa_spk_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	tosa_spk_func = ucontrol->value.integer.value[0];
-	tosa_ext_control(codec);
+	tosa_ext_control(&card->dapm);
 	return 1;
 }
 
@@ -191,24 +189,10 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
 
 	snd_soc_dapm_nc_pin(dapm, "OUT3");
 	snd_soc_dapm_nc_pin(dapm, "MONOOUT");
 
-	/* add tosa specific controls */
-	err = snd_soc_add_codec_controls(codec, tosa_controls,
-				ARRAY_SIZE(tosa_controls));
-	if (err < 0)
-		return err;
-
-	/* add tosa specific widgets */
-	snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets,
-				  ARRAY_SIZE(tosa_dapm_widgets));
-
-	/* set up tosa specific audio path audio_map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
 	return 0;
 }
 
@@ -239,6 +223,13 @@ static struct snd_soc_card tosa = {
 	.owner = THIS_MODULE,
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
+
+	.controls = tosa_controls,
+	.num_controls = ARRAY_SIZE(tosa_controls),
+	.dapm_widgets = tosa_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
 };
 
 static int tosa_probe(struct platform_device *pdev)