Merge tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been major modernization with the standard bus: in ALSA
  sequencer core and HD-audio.  Also, HD-audio receives the regmap
  support replacing the in-house cache register cache code.  These
  changes shouldn't impact the existing behavior, but rather
  refactoring.

  In addition, HD-audio got the code split to a core library part and
  the "legacy" driver parts.  This is a preliminary work for adapting
  the upcoming ASoC HD-audio driver, and the whole transition is still
  work in progress, likely finished in 4.1.

  Along with them, there are many updates in ASoC area as usual, too:
  lots of cleanups, Intel code shuffling, etc.

  Here are some highlights:

  ALSA core:
   - PCM: the audio timestamp / wallclock enhancement
   - PCM: fixes in DPCM management
   - Fixes / cleanups of user-space control element management
   - Sequencer: modernization using the standard bus

  HD-audio:
   - Modernization using the standard bus
   - Regmap support
   - Use standard runtime PM for codec power saving
   - Widget-path based power-saving for IDT, VIA and Realtek codecs
   - Reorganized sysfs entries for each codec object
   - More Dell headset support

  ASoC:
   - Move of jack registration to the card level
   - Lots of ASoC cleanups, mainly moving things from the CODEC level to
     the card level
   - Support for DAPM routes specified by both the machine driver and DT
   - Continuing improvements to rcar
   - pcm512x enhacements
   - Intel platforms updates
   - rt5670 updates / fixes
   - New platforms / devices: some non-DSP Qualcomm platforms, Google's
     Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC

  Misc:
   - ice1724: Improved ESI W192M support
   - emu10k1: Emu 1010 fixes/enhancement"

* tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
  ALSA: hda - set GET bit when adding a vendor verb to the codec regmap
  ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450
  ALSA: hda - Fix another race in runtime PM refcounting
  ALSA: hda - Expose codec type sysfs
  ALSA: ctl: fix to handle several elements added by one operation for userspace element
  ASoC: Intel: fix array_size.cocci warnings
  ASoC: n810: Automatically disconnect non-connected pins
  ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
  ASoC: davinci-evm: Use card DAPM context to access widgets
  ASoC: mop500_ab8500: Use card DAPM context to access widgets
  ASoC: wm1133-ev1: Use card DAPM context to access widgets
  ASoC: atmel: Improve machine driver compile test coverage
  ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
  ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_*
  ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate
  ASoC: rnsd: fix build regression without CONFIG_OF
  ALSA: emu10k1: add toggles for E-mu 1010 optical ports
  ALSA: ctl: fill identical information to return value when adding userspace elements
  ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls
  ALSA: ctl: confirm to return all identical information in 'activate' event
  ...

1 files changed, 430 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c
new file mode 100644
index 000000000000..49c09a0add79
--- /dev/null
+++ b/sound/soc/intel/boards/mfld_machine.c
@@ -0,0 +1,430 @@
+/*
+ *  mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/sn95031.h"
+
+#define MID_MONO 1
+#define MID_STEREO 2
+#define MID_MAX_CAP 5
+#define MFLD_JACK_INSERT 0x04
+
+enum soc_mic_bias_zones {
+	MFLD_MV_START = 0,
+	/* mic bias volutage range for Headphones*/
+	MFLD_MV_HP = 400,
+	/* mic bias volutage range for American Headset*/
+	MFLD_MV_AM_HS = 650,
+	/* mic bias volutage range for Headset*/
+	MFLD_MV_HS = 2000,
+	MFLD_MV_UNDEFINED,
+};
+
+static unsigned int	hs_switch;
+static unsigned int	lo_dac;
+static struct snd_soc_codec *mfld_codec;
+
+struct mfld_mc_private {
+	void __iomem *int_base;
+	u8 interrupt_status;
+};
+
+struct snd_soc_jack mfld_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mfld_jack_pins[] = {
+	{
+		.pin = "Headphones",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin = "AMIC1",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+/* jack detection voltage zones */
+static struct snd_soc_jack_zone mfld_zones[] = {
+	{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
+	{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
+};
+
+/* sound card controls */
+static const char *headset_switch_text[] = {"Earpiece", "Headset"};
+
+static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"};
+
+static const struct soc_enum headset_enum =
+	SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
+
+static const struct soc_enum lo_enum =
+	SOC_ENUM_SINGLE_EXT(4, lo_text);
+
+static int headset_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = hs_switch;
+	return 0;
+}
+
+static int headset_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+
+	if (ucontrol->value.integer.value[0] == hs_switch)
+		return 0;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	if (ucontrol->value.integer.value[0]) {
+		pr_debug("hs_set HS path\n");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+	} else {
+		pr_debug("hs_set EP path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
+	}
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	hs_switch = ucontrol->value.integer.value[0];
+
+	return 0;
+}
+
+static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm)
+{
+	snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT");
+	if (hs_switch) {
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+	} else {
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
+	}
+}
+
+static int lo_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = lo_dac;
+	return 0;
+}
+
+static int lo_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+
+	if (ucontrol->value.integer.value[0] == lo_dac)
+		return 0;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	/* we dont want to work with last state of lineout so just enable all
+	 * pins and then disable pins not required
+	 */
+	lo_enable_out_pins(dapm);
+
+	switch (ucontrol->value.integer.value[0]) {
+	case 0:
+		pr_debug("set vibra path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0);
+		break;
+
+	case 1:
+		pr_debug("set hs  path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22);
+		break;
+
+	case 2:
+		pr_debug("set spkr path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44);
+		break;
+
+	case 3:
+		pr_debug("set null path\n");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR");
+		snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66);
+		break;
+	}
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	lo_dac = ucontrol->value.integer.value[0];
+	return 0;
+}
+
+static const struct snd_kcontrol_new mfld_snd_controls[] = {
+	SOC_ENUM_EXT("Playback Switch", headset_enum,
+			headset_get_switch, headset_set_switch),
+	SOC_ENUM_EXT("Lineout Mux", lo_enum,
+			lo_get_switch, lo_set_switch),
+};
+
+static const struct snd_soc_dapm_widget mfld_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mfld_map[] = {
+	{"Headphones", NULL, "HPOUTR"},
+	{"Headphones", NULL, "HPOUTL"},
+	{"Mic", NULL, "AMIC1"},
+};
+
+static void mfld_jack_check(unsigned int intr_status)
+{
+	struct mfld_jack_data jack_data;
+
+	if (!mfld_codec)
+		return;
+
+	jack_data.mfld_jack = &mfld_jack;
+	jack_data.intr_id = intr_status;
+
+	sn95031_jack_detection(mfld_codec, &jack_data);
+	/* TODO: add american headset detection post gpiolib support */
+}
+
+static int mfld_init(struct snd_soc_pcm_runtime *runtime)
+{
+	struct snd_soc_dapm_context *dapm = &runtime->card->dapm;
+	int ret_val;
+
+	/* default is earpiece pin, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "Headphones");
+	/* default is lineout NC, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
+	lo_dac = 3;
+	hs_switch = 0;
+	/* we dont use linein in this so set to NC */
+	snd_soc_dapm_disable_pin(dapm, "LINEINL");
+	snd_soc_dapm_disable_pin(dapm, "LINEINR");
+
+	/* Headset and button jack detection */
+	ret_val = snd_soc_card_jack_new(runtime->card,
+			"Intel(R) MID Audio Jack", SND_JACK_HEADSET |
+			SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack,
+			mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins));
+	if (ret_val) {
+		pr_err("jack creation failed\n");
+		return ret_val;
+	}
+
+	ret_val = snd_soc_jack_add_zones(&mfld_jack,
+			ARRAY_SIZE(mfld_zones), mfld_zones);
+	if (ret_val) {
+		pr_err("adding jack zones failed\n");
+		return ret_val;
+	}
+
+	mfld_codec = runtime->codec;
+
+	/* we want to check if anything is inserted at boot,
+	 * so send a fake event to codec and it will read adc
+	 * to find if anything is there or not */
+	mfld_jack_check(MFLD_JACK_INSERT);
+	return ret_val;
+}
+
+static struct snd_soc_dai_link mfld_msic_dailink[] = {
+	{
+		.name = "Medfield Headset",
+		.stream_name = "Headset",
+		.cpu_dai_name = "Headset-cpu-dai",
+		.codec_dai_name = "SN95031 Headset",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = mfld_init,
+	},
+	{
+		.name = "Medfield Speaker",
+		.stream_name = "Speaker",
+		.cpu_dai_name = "Speaker-cpu-dai",
+		.codec_dai_name = "SN95031 Speaker",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Vibra",
+		.stream_name = "Vibra1",
+		.cpu_dai_name = "Vibra1-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra1",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Haptics",
+		.stream_name = "Vibra2",
+		.cpu_dai_name = "Vibra2-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra2",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Compress",
+		.stream_name = "Speaker",
+		.cpu_dai_name = "Compress-cpu-dai",
+		.codec_dai_name = "SN95031 Speaker",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_mfld = {
+	.name = "medfield_audio",
+	.owner = THIS_MODULE,
+	.dai_link = mfld_msic_dailink,
+	.num_links = ARRAY_SIZE(mfld_msic_dailink),
+
+	.controls = mfld_snd_controls,
+	.num_controls = ARRAY_SIZE(mfld_snd_controls),
+	.dapm_widgets = mfld_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mfld_widgets),
+	.dapm_routes = mfld_map,
+	.num_dapm_routes = ARRAY_SIZE(mfld_map),
+};
+
+static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
+{
+	struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
+
+	memcpy_fromio(&mc_private->interrupt_status,
+			((void *)(mc_private->int_base)),
+			sizeof(u8));
+	return IRQ_WAKE_THREAD;
+}
+
+static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
+{
+	struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
+
+	mfld_jack_check(mc_drv_ctx->interrupt_status);
+
+	return IRQ_HANDLED;
+}
+
+static int snd_mfld_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0, irq;
+	struct mfld_mc_private *mc_drv_ctx;
+	struct resource *irq_mem;
+
+	pr_debug("snd_mfld_mc_probe called\n");
+
+	/* retrive the irq number */
+	irq = platform_get_irq(pdev, 0);
+
+	/* audio interrupt base of SRAM location where
+	 * interrupts are stored by System FW */
+	mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
+	if (!mc_drv_ctx) {
+		pr_err("allocation failed\n");
+		return -ENOMEM;
+	}
+
+	irq_mem = platform_get_resource_byname(
+				pdev, IORESOURCE_MEM, "IRQ_BASE");
+	if (!irq_mem) {
+		pr_err("no mem resource given\n");
+		return -ENODEV;
+	}
+	mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start,
+						    resource_size(irq_mem));
+	if (!mc_drv_ctx->int_base) {
+		pr_err("Mapping of cache failed\n");
+		return -ENOMEM;
+	}
+	/* register for interrupt */
+	ret_val = devm_request_threaded_irq(&pdev->dev, irq,
+			snd_mfld_jack_intr_handler,
+			snd_mfld_jack_detection,
+			IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
+	if (ret_val) {
+		pr_err("cannot register IRQ\n");
+		return ret_val;
+	}
+	/* register the soc card */
+	snd_soc_card_mfld.dev = &pdev->dev;
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
+	if (ret_val) {
+		pr_debug("snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, mc_drv_ctx);
+	pr_debug("successfully exited probe\n");
+	return 0;
+}
+
+static struct platform_driver snd_mfld_mc_driver = {
+	.driver = {
+		.name = "msic_audio",
+	},
+	.probe = snd_mfld_mc_probe,
+};
+
+module_platform_driver(snd_mfld_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:msic-audio");