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, 292 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
new file mode 100644
index 000000000000..8bafaf6ceab1
--- /dev/null
+++ b/sound/soc/intel/boards/broadwell.c
@@ -0,0 +1,292 @@
+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * 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 <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+
+#include "../common/sst-dsp.h"
+#include "../haswell/sst-haswell-ipc.h"
+
+#include "../../codecs/rt286.h"
+
+static struct snd_soc_jack broadwell_headset;
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin broadwell_headset_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headphone Jack",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static const struct snd_kcontrol_new broadwell_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+};
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("DMIC1", NULL),
+	SND_SOC_DAPM_MIC("DMIC2", NULL),
+	SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack deteck */
+	{"Headphone Jack", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+	{"LINE1", NULL, "Line Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC1"},
+	{"DMIC2 Pin", NULL, "DMIC2"},
+
+	/* CODEC BE connections */
+	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret = 0;
+	ret = snd_soc_card_jack_new(rtd->card, "Headset",
+		SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset,
+		broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins));
+	if (ret)
+		return ret;
+
+	rt286_mic_detect(codec, &broadwell_headset);
+	return 0;
+}
+
+
+static int broadwell_ssp0_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 ADSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+static int broadwell_rt286_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;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+		SND_SOC_CLOCK_IN);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+	.hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+	struct sst_hsw *broadwell = pdata->dsp;
+	int ret;
+
+	/* Set ADSP SSP port settings */
+	ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+		SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+		SST_HSW_DEVICE_CLOCK_MASTER, 9);
+	if (ret < 0) {
+		dev_err(rtd->dev, "error: failed to set device config\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "System PCM",
+		.stream_name = "System Playback/Capture",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.init = broadwell_rtd_init,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "Offload0",
+		.stream_name = "Offload0 Playback",
+		.cpu_dai_name = "Offload0 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload1",
+		.stream_name = "Offload1 Playback",
+		.cpu_dai_name = "Offload1 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Loopback PCM",
+		.stream_name = "Loopback",
+		.cpu_dai_name = "Loopback Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 0,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "Codec",
+		.be_id = 0,
+		.cpu_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "i2c-INT343A:00",
+		.codec_dai_name = "rt286-aif1",
+		.init = broadwell_rt286_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = broadwell_ssp0_fixup,
+		.ops = &broadwell_rt286_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+static int broadwell_suspend(struct snd_soc_card *card){
+	struct snd_soc_codec *codec;
+
+	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+		if (!strcmp(codec->component.name, "i2c-INT343A:00")) {
+			dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
+			rt286_mic_detect(codec, NULL);
+			break;
+		}
+	}
+	return 0;
+}
+
+static int broadwell_resume(struct snd_soc_card *card){
+	struct snd_soc_codec *codec;
+
+	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+		if (!strcmp(codec->component.name, "i2c-INT343A:00")) {
+			dev_dbg(codec->dev, "enabling jack detect for resume.\n");
+			rt286_mic_detect(codec, &broadwell_headset);
+			break;
+		}
+	}
+	return 0;
+}
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+	.name = "broadwell-rt286",
+	.owner = THIS_MODULE,
+	.dai_link = broadwell_rt286_dais,
+	.num_links = ARRAY_SIZE(broadwell_rt286_dais),
+	.controls = broadwell_controls,
+	.num_controls = ARRAY_SIZE(broadwell_controls),
+	.dapm_widgets = broadwell_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+	.dapm_routes = broadwell_rt286_map,
+	.num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+	.fully_routed = true,
+	.suspend_pre = broadwell_suspend,
+	.resume_post = broadwell_resume,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+	broadwell_rt286.dev = &pdev->dev;
+
+	return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&broadwell_rt286);
+	return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+	.probe = broadwell_audio_probe,
+	.remove = broadwell_audio_remove,
+	.driver = {
+		.name = "broadwell-audio",
+	},
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");