Merge tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Changes are seen in a wide range of codes, mainly due to ASoC DAPM
  requirements; HD-audio shows a high peak in diffstat, it's just a
  removal of bunch of old static quirks.

  Some highlights:

   - HDPM: Updates for AIO/RayDAT support, TCO/sync support

   - RME96: Add PCM sync support

   - HD-audio:

     * A few HDMI/DP audio updates (CA assignment fix, stream switching
       fix, Intel DP device list support)
     * Device specific fixes (ASUS/CXT HP mic support, Thinkpad mic
       improvements, Chromebook fixes, STAC9228 Dell fixes)
     * Replace the all static quirks for AD codecs with the generic
       parser
     * WAKEEN support for handling irqs in the power saving mode

   - USB-audio: Clean up implicit fb handling and related codes

   - DAPM is now mandatory for ASoC CODEC drivers; all existing drivers
     have had some level of DAPM support added.  In addition, a lot of
     cleanups and improvements in DAPM.

   - Support for ASoC cross-platform compile test

   - New drivers and support for Analog Devices ADAU1702 and
     ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and
     WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas
     R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and
     PCM1792A and Wolfson Microelectronics WM8997

   - DT bindings for kirkwood and i.MX S/PDIF

   - Clean up and bug fixes: ssm2602, rt5640 and sgtl5000.

   - Core helpers for bitbanged AC'97 reset"

* tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (375 commits)
  ALSA: hda - Re-setup HDMI pin and audio infoframe on stream switches
  ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA
  ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional
  ASoC: soc-pcm: Allow to specify unidirectional dai_link
  ASoC: fsl_spdif: Staticse non-exported symbols
  ASoC: ssm2602: Fix cache sync
  ASoC: Remove unused sysfs_registered field from snd_soc_codec struct
  ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct
  ASoC: Remove unused control_type field from snd_soc_codec struct
  ASoC: fsl: Add one blank space after ':=' in Makefile
  ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c
  ASoC: rt5640: change widget sequence for depop
  ASoC: dapm: Fix auto-disable for inverted controls
  ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF
  ASoC: Samsung: Do not queue cyclic buffers multiple times
  ASoC: ep93xx-i2s: Remove unnecessary dev_set_drvdata()
  ASoC: designware_i2s: Remove unnecessary dev_set_drvdata()
  ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe()
  ASoC: fsl: Add S/PDIF machine driver
  ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards
  ...

1 files changed, 78 insertions, 1 deletions

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index de00ce166470..4edd2d0f9a3c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -66,6 +66,8 @@ struct conexant_spec {
 	hda_nid_t eapds[4];
 	bool dynamic_eapd;
 
+	unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+
 #ifdef ENABLE_CXT_STATIC_QUIRKS
 	const struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
@@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	snd_hda_gen_init(codec);
 	if (!spec->dynamic_eapd)
 		cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
+
 	return 0;
 }
 
@@ -3224,6 +3229,8 @@ enum {
 	CXT_PINCFG_LEMOTE_A1205,
 	CXT_FIXUP_STEREO_DMIC,
 	CXT_FIXUP_INC_MIC_BOOST,
+	CXT_FIXUP_HEADPHONE_MIC_PIN,
+	CXT_FIXUP_HEADPHONE_MIC,
 };
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec,
 				  (0 << AC_AMPCAP_MUTE_SHIFT));
 }
 
+static void cxt_update_headset_mode(struct hda_codec *codec)
+{
+	/* The verbs used in this function were tested on a Conexant CX20751/2 codec. */
+	int i;
+	bool mic_mode = false;
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+	hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+
+	for (i = 0; i < cfg->num_inputs; i++)
+		if (cfg->inputs[i].pin == mux_pin) {
+			mic_mode = !!cfg->inputs[i].is_headphone_mic;
+			break;
+		}
+
+	if (mic_mode) {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = false;
+	} else {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]);
+	}
+
+	snd_hda_gen_update_outputs(codec);
+}
+
+static void cxt_update_headset_mode_hook(struct hda_codec *codec,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	cxt_update_headset_mode(codec);
+}
+
+static void cxt_fixup_headphone_mic(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC;
+		break;
+	case HDA_FIXUP_ACT_PROBE:
+		spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
+		spec->gen.automute_hook = cxt_update_headset_mode;
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		cxt_update_headset_mode(codec);
+		break;
+	}
+}
+
+
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt5066_increase_mic_boost,
 	},
+	[CXT_FIXUP_HEADPHONE_MIC_PIN] = {
+		.type = HDA_FIXUP_PINS,
+		.chained = true,
+		.chain_id = CXT_FIXUP_HEADPHONE_MIC,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+			{ }
+		}
+	},
+	[CXT_FIXUP_HEADPHONE_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_headphone_mic,
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL,
+				       spec->parse_flags);
 	if (err < 0)
 		goto error;
 
@@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		codec->bus->allow_bus_reset = 1;
 	}
 
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error: