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, 324 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
new file mode 100644
index 000000000000..20a28b22e30f
--- /dev/null
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -0,0 +1,324 @@
+/*
+ *  cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
+ *                     Cherrytrail and Braswell, with RT5645 codec.
+ *
+ *  Copyright (C) 2015 Intel Corp
+ *  Author: Fang, Yang A <yang.a.fang@intel.com>
+ *	        N,Harshapriya <harshapriya.n@intel.com>
+ *  This file is modified from cht_bsw_rt5672.c
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5645.h"
+#include "../atom/sst-atom-controls.h"
+
+#define CHT_PLAT_CLK_3_HZ	19200000
+#define CHT_CODEC_DAI	"rt5645-aif1"
+
+struct cht_mc_private {
+	struct snd_soc_jack hp_jack;
+	struct snd_soc_jack mic_jack;
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+	int i;
+
+	for (i = 0; i < card->num_rtd; i++) {
+		struct snd_soc_pcm_runtime *rtd;
+
+		rtd = card->rtd + i;
+		if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+			     strlen(CHT_CODEC_DAI)))
+			return rtd->codec_dai;
+	}
+	return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret;
+
+	codec_dai = cht_get_codec_dai(card);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+		return -EIO;
+	}
+
+	if (!SND_SOC_DAPM_EVENT_OFF(event))
+		return 0;
+
+	/* Set codec sysclk source to its internal clock because codec PLL will
+	 * be off when idle and MCLK will also be off by ACPI when codec is
+	 * runtime suspended. Codec needs clock for jack detection and button
+	 * press.
+	 */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
+			0, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+	{"IN1P", NULL, "Headset Mic"},
+	{"IN1N", NULL, "Headset Mic"},
+	{"DMIC L1", NULL, "Int Mic"},
+	{"DMIC R1", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Ext Spk", NULL, "SPOL"},
+	{"Ext Spk", NULL, "SPOR"},
+	{"AIF1 Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx" },
+	{"codec_in1", NULL, "ssp2 Rx" },
+	{"ssp2 Rx", NULL, "AIF1 Capture"},
+	{"Headphone", NULL, "Platform Clock"},
+	{"Headset Mic", NULL, "Platform Clock"},
+	{"Int Mic", NULL, "Platform Clock"},
+	{"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
+				  CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
+				params_rate(params) * 512, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_codec *codec = runtime->codec;
+	struct snd_soc_dai *codec_dai = runtime->codec_dai;
+	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
+
+	/* Select clk_i2s1_asrc as ASRC clock source */
+	rt5645_sel_asrc_clk_src(codec,
+				RT5645_DA_STEREO_FILTER |
+				RT5645_DA_MONO_L_FILTER |
+				RT5645_DA_MONO_R_FILTER |
+				RT5645_AD_STEREO_FILTER,
+				RT5645_CLK_SEL_I2S1_ASRC);
+
+	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+	if (ret < 0) {
+		dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &ctx->hp_jack,
+				    NULL, 0);
+	if (ret) {
+		dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_card_jack_new(runtime->card, "Mic Jack",
+				    SND_JACK_MICROPHONE, &ctx->mic_jack,
+				    NULL, 0);
+	if (ret) {
+		dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack);
+
+	return ret;
+}
+
+static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
+static unsigned int rates_48000[] = {
+	48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+	.startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+	.hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.ignore_suspend = 1,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Compressed Port",
+		.stream_name = "Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5645-aif1",
+		.codec_name = "i2c-10EC5645:00",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+					| SND_SOC_DAIFMT_CBS_CFS,
+		.init = cht_codec_init,
+		.be_hw_params_fixup = cht_codec_fixup,
+		.ignore_suspend = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &cht_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+	.name = "chtrt5645",
+	.dai_link = cht_dailink,
+	.num_links = ARRAY_SIZE(cht_dailink),
+	.dapm_widgets = cht_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+	.dapm_routes = cht_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+	.controls = cht_mc_controls,
+	.num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	struct cht_mc_private *drv;
+
+	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
+	if (!drv)
+		return -ENOMEM;
+
+	snd_soc_card_cht.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &snd_soc_card_cht);
+	return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+	.driver = {
+		.name = "cht-bsw-rt5645",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
+MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5645");