Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound update from Takashi Iwai:
 "This is a fairly quiet release in all sound area.  Only a little bit
  of changes in the core side while most of changes are seen in the
  drivers.

  HD-audio:
   - A few new codec additions for Nvidia, Realtek and VIA
   - Intel Haswell audio support
   - Support for "phantom" jacks for consistent jack reporting
   - Major clean-ups in HDMI/DP driver codes
   - A workaround for inverted digital-mic pins with Realtek codecs
   - Removal of beep_mode=2 option

  ASoC:
   - Added the ability to add and remove DAPM paths dynamically, mostly
     for reparenting on clock changes
   - New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
     reference platform and ttc-dkp
   - New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
     Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
   - New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500,
     TI Isabelle and Wolfson Microelectronics WM5102 and WM5110
   - DAPM fixes for the recent locking changes
   - Fix for _PRE and _POST widgets (which have been broken for a few
     releases now)
   - A couple of minor driver updates

  Misc
   - Conversion to new dev_pm_ops in platform and PCI drivers
   - LTC support and some fixes in PCXHR driver
   - A few fixes and PM support for ISA OPti9xx and WSS cards
   - Some TLV code cleanup
   - Move driver-specific headers from include/sound to local dirs"

* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (212 commits)
  ASoC: dapm: Fix _PRE and _POST events for DAPM performance improvements
  ALSA: hda - add dock support for Thinkpad X230 Tablet
  ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
  ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define
  ASoC: littlemill: Add userspace control of the WM1250 I/O
  ASoC: wm8994: Update micdet for irqdomain conversion
  ALSA: hda - make sure alc268 does not OOPS on codec parse
  ALSA: hda - Add support for Realtek ALC282
  ALSA: hda - Fix index number conflicts of phantom jacks
  ALSA: opti9xx: Fix section mismatch by PM support
  ALSA: snd-opti9xx: Implement suspend/resume
  ALSA: hda - Add new GPU codec ID to snd-hda
  ALSA: hda - Fix driver type of Haswell controller to AZX_DRIVER_SCH
  ALSA: hda - add Haswell HDMI codec id
  ALSA: hda - Add DeviceID for Haswell HDA
  ALSA: wss_lib: Fix resume on Yamaha OPL3-SAx
  ALSA: wss_lib: fix suspend/resume
  ALSA: es1938: replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE
  ALSA: tlv: add DECLARE_TLV_DB_RANGE()
  ALSA: tlv: add DECLARE_TLV_CONTAINER()
  ...

1 files changed, 173 insertions, 0 deletions

diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 000000000000..935491a8a770
--- /dev/null
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,173 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+	return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+	 .name = "88pm860x i2s",
+	 .stream_name = "audio playback",
+	 .codec_name = "88pm860x-codec",
+	 .platform_name = "mmp-pcm-audio",
+	 .cpu_dai_name = "pxa-ssp-dai.1",
+	 .codec_dai_name = "88pm860x-i2s",
+	 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM,
+	 .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+	.name = "ttc-dkb-hifi",
+	.dai_link = ttc_pm860x_hifi_dai,
+	.num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+	.dapm_widgets = ttc_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+	.dapm_routes = ttc_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int __devinit ttc_dkb_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &ttc_dkb_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit ttc_dkb_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver ttc_dkb_driver = {
+	.driver		= {
+		.name	= "ttc-dkb-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= ttc_dkb_probe,
+	.remove		= __devexit_p(ttc_dkb_remove),
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");