Merge tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This time it's a relatively calm update batch, but the amount isn't
  too small in the end.  Here we go over some highlights:

  ALSA core:
   - One major change is the support of nonatomic PCM operations.  This
     allows the trigger and other callbacks to call schedule(), which
     would be useful for mailbox type communications.  Already some
     drivers (Digigram ones) have been converted to use together with
     threaded irqs as an example.
   - Improvement / fixes of DSD PCM format support

  HD-audio:
   - Large volume of rewrites are found in Realtek codec driver for
     converting Dell and HP quirks to generic forms.
   - Inverted dmic code cleanup from David.
   - Realtek COEF access has been optimized.
   - Now HD-audio jack infrastructure allows multiple callbacks, which
     fixes / simplifies the jack-dependent power controls on STAC/IDT
     and VIA codecs.
   - Many additional device-specific fixups as usual
   - A few deadcode cleanups, CA0132 code cleanup, etc.

  ASoC:
   - More componentization work from Lars-Peter, this time mainly
     cleaning up the suspend and bias level transition callbacks.
   - Real system support for the Intel drivers and a bunch of fixes and
     enhancements for the associated CODEC drivers, this is going to
     need a lot quirks over time due to the lack of any firmware
     description of the boards.
   - Jack detect support for simple card from Dylan Reid.
   - A bunch of small fixes and enhancements for the Freescale drivers.
   - New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32,
     Everest Semiconductor ES8328 and Freescale cards using the ASRC in
     newer i.MX processors.
   - A few simple-card fixes, mostly cleanups but also a fix for
     interaction between GPIO 0 and simple-card.

  Misc:
   - Virtuoso / Oxygen updates by Clemens
   - USB-audio: Yamaha MOTIF XF MIDI port name fixes
   - Conversion of kernel messages to standard dev_*() in ctxfi driver"

* tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (251 commits)
  ASoC: mc13783: Ensure we only try to dereference valid of_nodes
  ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_txctrl
  ALSA: hda - Add dock port support to Thinkpad L440 (71aa:501e)
  ALSA: Allow pass NULL dev for snd_pci_quirk_lookup()
  ASoC: imx-es8328: Fix of_node_put() call with uninitialized object
  ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb()
  ASoC: simple-card: Initialize headphone and mic GPIO numbers
  ASoC: imx-es8328: Fix missing return code in imx_es8328_probe()
  ALSA: hda - Add dock support for Thinkpad T440 (17aa:2212)
  ALSA: usb: caiaq: check for cdev->n_streams > 1
  ASoC: 88pm860x-codec: Fix possibly missing string termination
  ASoC: core: fix use after free in snd_soc_remove_platform()
  ASoC: soc-dapm: fix use after free
  ALSA: hda - Make the inv dmic handling for Realtek use generic parser
  ALSA: hda - Add Inverted Internal mic for Samsung Ativ book 9 (NP900X3G)
  ALSA: hda - Add inverted internal mic for Asus Aspire 4830T
  ASoC: Intel: byt-rt5640: fix coccinelle warnings
  ASoC: fsl_esai doc: Add "fsl,vf610-esai" as compatible string
  ASoC: da732x: Remove unnecessary KERN_ERR in pr_err()
  ASoC: simple-card: Fix detect gpio documentation.
  ...

1 files changed, 50 insertions, 20 deletions

diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 23b32960ff1d..f039dc825971 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -78,6 +78,44 @@ struct tas2552_data {
 	unsigned int mclk;
 };
 
+/* Input mux controls */
+static const char *tas2552_input_texts[] = {
+	"Digital", "Analog"
+};
+
+static SOC_ENUM_SINGLE_DECL(tas2552_input_mux_enum, TAS2552_CFG_3, 7,
+			    tas2552_input_texts);
+
+static const struct snd_kcontrol_new tas2552_input_mux_control[] = {
+	SOC_DAPM_ENUM("Input selection", tas2552_input_mux_enum)
+};
+
+static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] =
+{
+	SND_SOC_DAPM_INPUT("IN"),
+
+	/* MUX Controls */
+	SND_SOC_DAPM_MUX("Input selection", SND_SOC_NOPM, 0, 0,
+				tas2552_input_mux_control),
+
+	SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("OUT")
+};
+
+static const struct snd_soc_dapm_route tas2552_audio_map[] = {
+	{"DAC", NULL, "DAC IN"},
+	{"Input selection", "Digital", "DAC"},
+	{"Input selection", "Analog", "IN"},
+	{"ClassD", NULL, "Input selection"},
+	{"OUT", NULL, "ClassD"},
+	{"ClassD", NULL, "PLL"},
+};
+
+#ifdef CONFIG_PM_RUNTIME
 static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
 {
 	u8 cfg1_reg;
@@ -90,6 +128,7 @@ static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
 	snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1,
 						 TAS2552_SWS_MASK, cfg1_reg);
 }
+#endif
 
 static int tas2552_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params,
@@ -101,10 +140,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream,
 	int d;
 	u8 p, j;
 
-	/* Turn on Class D amplifier */
-	snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK,
-						TAS2552_CLASSD_EN);
-
 	if (!tas2552->mclk)
 		return -EINVAL;
 
@@ -147,9 +182,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream,
 
 	}
 
-	snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE,
-						TAS2552_PLL_ENABLE);
-
 	return 0;
 }
 
@@ -269,19 +301,10 @@ static const struct dev_pm_ops tas2552_pm = {
 			   NULL)
 };
 
-static void tas2552_shutdown(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
-}
-
 static struct snd_soc_dai_ops tas2552_speaker_dai_ops = {
 	.hw_params	= tas2552_hw_params,
 	.set_sysclk	= tas2552_set_dai_sysclk,
 	.set_fmt	= tas2552_set_dai_fmt,
-	.shutdown	= tas2552_shutdown,
 	.digital_mute = tas2552_mute,
 };
 
@@ -294,7 +317,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
 	{
 		.name = "tas2552-amplifier",
 		.playback = {
-			.stream_name = "Speaker",
+			.stream_name = "Playback",
 			.channels_min = 2,
 			.channels_max = 2,
 			.rates = SNDRV_PCM_RATE_8000_192000,
@@ -312,6 +335,7 @@ static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24);
 static const struct snd_kcontrol_new tas2552_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Driver Playback Volume",
 			 TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv),
+	SOC_DAPM_SINGLE("Playback AMP", SND_SOC_NOPM, 0, 1, 0),
 };
 
 static const struct reg_default tas2552_init_regs[] = {
@@ -321,6 +345,7 @@ static const struct reg_default tas2552_init_regs[] = {
 static int tas2552_codec_probe(struct snd_soc_codec *codec)
 {
 	struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	tas2552->codec = codec;
@@ -362,9 +387,14 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec)
 		goto patch_fail;
 	}
 
-	snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN |
-				  TAS2552_BOOST_EN | TAS2552_APT_EN |
-				  TAS2552_LIM_EN);
+	snd_soc_write(codec, TAS2552_CFG_2, TAS2552_BOOST_EN |
+				  TAS2552_APT_EN | TAS2552_LIM_EN);
+
+	snd_soc_dapm_new_controls(dapm, tas2552_dapm_widgets,
+				ARRAY_SIZE(tas2552_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, tas2552_audio_map,
+				ARRAY_SIZE(tas2552_audio_map));
+
 	return 0;
 
 patch_fail: