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, 57 insertions, 111 deletions

diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 98a04c11202d..b0800337b79e 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -192,44 +192,6 @@ static struct snd_soc_ops neo1973_voice_ops = {
 	.hw_free = neo1973_voice_hw_free,
 };
 
-/* Shared routes and controls */
-
-static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
-	SND_SOC_DAPM_LINE("GSM Line In", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Handset Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
-	/* Connections to the GSM Module */
-	{"GSM Line Out", NULL, "MONO1"},
-	{"GSM Line Out", NULL, "MONO2"},
-	{"RXP", NULL, "GSM Line In"},
-	{"RXN", NULL, "GSM Line In"},
-
-	/* Connections to Headset */
-	{"MIC1", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Headset Mic"},
-
-	/* Call Mic */
-	{"MIC2", NULL, "Mic Bias"},
-	{"MIC2N", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Handset Mic"},
-
-	/* Connect the ALC pins */
-	{"ACIN", NULL, "ACOP"},
-};
-
-static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
-	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
-	SOC_DAPM_PIN_SWITCH("GSM Line In"),
-	SOC_DAPM_PIN_SWITCH("Headset Mic"),
-	SOC_DAPM_PIN_SWITCH("Handset Mic"),
-};
-
-/* GTA02 specific routes and controls */
-
 static int gta02_speaker_enabled;
 
 static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
@@ -257,7 +219,34 @@ static int lm4853_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
+static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line In", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Handset Mic", NULL),
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+};
+
+static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
+	/* Connections to the GSM Module */
+	{"GSM Line Out", NULL, "MONO1"},
+	{"GSM Line Out", NULL, "MONO2"},
+	{"RXP", NULL, "GSM Line In"},
+	{"RXN", NULL, "GSM Line In"},
+
+	/* Connections to Headset */
+	{"MIC1", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Headset Mic"},
+
+	/* Call Mic */
+	{"MIC2", NULL, "Mic Bias"},
+	{"MIC2N", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Handset Mic"},
+
+	/* Connect the ALC pins */
+	{"ACIN", NULL, "ACOP"},
+
 	/* Connections to the amp */
 	{"Stereo Out", NULL, "LOUT1"},
 	{"Stereo Out", NULL, "ROUT1"},
@@ -267,7 +256,11 @@ static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
 	{"Handset Spk", NULL, "ROUT2"},
 };
 
-static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
+static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
+	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+	SOC_DAPM_PIN_SWITCH("GSM Line In"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Handset Mic"),
 	SOC_DAPM_PIN_SWITCH("Handset Spk"),
 	SOC_DAPM_PIN_SWITCH("Stereo Out"),
 
@@ -276,86 +269,32 @@ static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
 		lm4853_set_spk),
 };
 
-static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_SPK("Handset Spk", NULL),
-	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
-};
-
-static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
-
-	ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets,
-			ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets));
-	if (ret)
-		return ret;
-
-	ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes,
-			ARRAY_SIZE(neo1973_gta02_routes));
-	if (ret)
-		return ret;
-
-	ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls,
-			ARRAY_SIZE(neo1973_gta02_wm8753_controls));
-	if (ret)
-		return ret;
-
-	snd_soc_dapm_disable_pin(dapm, "Stereo Out");
-	snd_soc_dapm_disable_pin(dapm, "Handset Spk");
-	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
-
-	return 0;
-}
-
 static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
+	struct snd_soc_card *card = rtd->card;
 
 	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(dapm, "OUT3");
-	snd_soc_dapm_nc_pin(dapm, "OUT4");
-	snd_soc_dapm_nc_pin(dapm, "LINE1");
-	snd_soc_dapm_nc_pin(dapm, "LINE2");
-
-	/* Add neo1973 specific widgets */
-	ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets,
-			ARRAY_SIZE(neo1973_wm8753_dapm_widgets));
-	if (ret)
-		return ret;
-
-	/* add neo1973 specific controls */
-	ret = snd_soc_add_card_controls(rtd->card, neo1973_wm8753_controls,
-			ARRAY_SIZE(neo1973_wm8753_controls));
-	if (ret)
-		return ret;
-
-	/* set up neo1973 specific audio routes */
-	ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes,
-			ARRAY_SIZE(neo1973_wm8753_routes));
-	if (ret)
-		return ret;
+	snd_soc_dapm_nc_pin(&codec->dapm, "OUT3");
+	snd_soc_dapm_nc_pin(&codec->dapm, "OUT4");
+	snd_soc_dapm_nc_pin(&codec->dapm, "LINE1");
+	snd_soc_dapm_nc_pin(&codec->dapm, "LINE2");
 
 	/* set endpoints to default off mode */
-	snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-	snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-	snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+	snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out");
+	snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In");
+	snd_soc_dapm_disable_pin(&card->dapm, "Headset Mic");
+	snd_soc_dapm_disable_pin(&card->dapm, "Handset Mic");
+	snd_soc_dapm_disable_pin(&card->dapm, "Stereo Out");
+	snd_soc_dapm_disable_pin(&card->dapm, "Handset Spk");
 
 	/* allow audio paths from the GSM modem to run during suspend */
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
-	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
-
-	if (machine_is_neo1973_gta02()) {
-		ret = neo1973_gta02_wm8753_init(codec);
-		if (ret)
-			return ret;
-	}
+	snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line Out");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line In");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Mic");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Spk");
 
 	return 0;
 }
@@ -409,6 +348,13 @@ static struct snd_soc_card neo1973 = {
 	.num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
 	.codec_conf = neo1973_codec_conf,
 	.num_configs = ARRAY_SIZE(neo1973_codec_conf),
+
+	.controls = neo1973_wm8753_controls,
+	.num_controls = ARRAY_SIZE(neo1973_wm8753_controls),
+	.dapm_widgets = neo1973_wm8753_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets),
+	.dapm_routes = neo1973_wm8753_routes,
+	.num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes),
 };
 
 static struct platform_device *neo1973_snd_device;