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, 227 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
new file mode 100644
index 000000000000..7f55d59024a8
--- /dev/null
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -0,0 +1,227 @@
+/*
+ *  byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform
+ *
+ *  Copyright (C) 2014 Intel Corp
+ *  Author: Subhransu S. Prusty <subhransu.s.prusty@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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/input.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/rt5640.h"
+#include "../atom/sst-atom-controls.h"
+
+static const struct snd_soc_dapm_widget byt_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),
+};
+
+static const struct snd_soc_dapm_route byt_audio_map[] = {
+	{"IN2P", NULL, "Headset Mic"},
+	{"IN2N", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "MICBIAS1"},
+	{"IN1P", NULL, "MICBIAS1"},
+	{"LDO2", NULL, "Int Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Ext Spk", NULL, "SPOLP"},
+	{"Ext Spk", NULL, "SPOLN"},
+	{"Ext Spk", NULL, "SPORP"},
+	{"Ext Spk", NULL, "SPORN"},
+
+	{"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"},
+};
+
+static const struct snd_kcontrol_new byt_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 byt_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;
+
+	snd_soc_dai_set_bclk_ratio(codec_dai, 50);
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
+				     params_rate(params) * 512,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec clock %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+				  params_rate(params) * 50,
+				  params_rate(params) * 512);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_pcm_stream byt_dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	.rate_min = 48000,
+	.rate_max = 48000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static int byt_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 byt_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 byt_aif1_ops = {
+	.startup = byt_aif1_startup,
+};
+
+static struct snd_soc_ops byt_be_ssp2_ops = {
+	.hw_params = byt_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link byt_dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Baytrail Audio Port",
+		.stream_name = "Baytrail 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 = &byt_aif1_ops,
+	},
+	[MERR_DPCM_COMPR] = {
+		.name = "Baytrail Compressed Port",
+		.stream_name = "Baytrail Compress",
+		.cpu_dai_name = "compress-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+	},
+		/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.be_id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "rt5640-aif1",
+		.codec_name = "i2c-10EC5640:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = byt_codec_fixup,
+		.ignore_suspend = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &byt_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_byt = {
+	.name = "baytrailcraudio",
+	.dai_link = byt_dailink,
+	.num_links = ARRAY_SIZE(byt_dailink),
+	.dapm_widgets = byt_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets),
+	.dapm_routes = byt_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(byt_audio_map),
+	.controls = byt_mc_controls,
+	.num_controls = ARRAY_SIZE(byt_mc_controls),
+};
+
+static int snd_byt_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+
+	/* register the soc card */
+	snd_soc_card_byt.dev = &pdev->dev;
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt);
+	if (ret_val) {
+		dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &snd_soc_card_byt);
+	return ret_val;
+}
+
+static struct platform_driver snd_byt_mc_driver = {
+	.driver = {
+		.name = "bytt100_rt5640",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = snd_byt_mc_probe,
+};
+
+module_platform_driver(snd_byt_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
+MODULE_AUTHOR("Subhransu S. Prusty <subhransu.s.prusty@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytt100_rt5640");