Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've had quite busy weeks in this cycle.  Looking at ALSA core, the
  significant changes are a few fixes wrt timer and sequencer ioctls
  that have been revealed by fuzzer recently.  Other than that, ASoC
  core got a few updates about DAI link handling, but these are rather
  straightforward refactoring.

  In drivers scene, ASoC received quite lots of new drivers in addition
  to bunch of updates for still ongoing Intel Skylake support and
  topology API.  HD-audio gained a new HDMI/DP hotplug notification via
  component.  FireWire got a pile of code refactoring/updates with
  SCS.1x driver integration.

  More highlights are shown below.

  [ NOTE: this contains also many commits for DRM.  This is due to the
    pull of drm stable branch into sound tree, as the base of i915 audio
    component work for HD-audio.  The highlights below don't contain
    these DRM changes, as these are supposed to be pulled via drm tree
    in anyway sooner or later.  ]

  Core:
   - Handful fixes to harden ALSA timer and sequencer ioctls against
     races reported by syzkaller fuzzer
   - Irq description string can be unique to each card; only for
     HD-audio for now

  ASoC:
   - Conversion of the array of DAI links to a list for supporting
     dynamically adding and removing DAI links
   - Topology API enhancements to make everything more component based
     and being able to specify PCM links via topology
   - Some more fixes for the topology code, though it is still not final
     and ready for enabling in production; we really need to get to the
     point where that can be done
   - A pile of changes for Intel SkyLake drivers which hopefully deliver
     some useful initial functionality for systems with this chipset,
     though there is more work still to come
   - Lots of new features and cleanups for the Renesas drivers
   - ANC support for WM5110
   - New drivers: Imagination Technologies IPs, Atmel class D speaker,
     Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
     RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
   - Rename PCM1792a driver to be generic pcm179x

  HD-Audio:
   - Use audio component for i915 HDMI/DP hotplug handling
   - On-demand binding with i915 driver
   - bdl_pos_adj parameter adjustment for Baytrail controllers
   - Enable power_save_node for CX20722; this shouldn't lead to
     regression, hopefully
   - Kabylake HDMI/DP codec support
   - Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
     machines
   - A few code refactoring

  FireWire:
   - Lots of code cleanup and refactoring
   - Integrate the support of SCS.1x devices into snd-oxfw driver;
     snd-scs1x driver is obsoleted

  USB-audio:
   - Fix possible NULL dereference at disconnection
   - A regression fix for Native Instruments devices

  Misc:
   - A few code cleanups of fm801 driver"

* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
  ALSA: timer: Code cleanup
  ALSA: timer: Harden slave timer list handling
  ALSA: hda - Add fixup for Dell Latitidue E6540
  ALSA: timer: Fix race among timer ioctls
  ALSA: hda - add codec support for Kabylake display audio codec
  ALSA: timer: Fix double unlink of active_list
  ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
  ALSA: hda - fix the headset mic detection problem for a Dell laptop
  ALSA: hda - Fix white noise on Dell Latitude E5550
  ALSA: hda_intel: add card number to irq description
  ALSA: seq: Fix race at timer setup and close
  ALSA: seq: Fix missing NULL check at remove_events ioctl
  ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
  ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
  ASoC: AMD: Add missing include file
  ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
  ALSA: usb: Add native DSD support for Oppo HA-1
  ASoC: Make aux_dev more like a generic component
  ASoC: bcm2835: cleanup includes by ordering them alphabetically
  ASoC: AMD: Manage ACP 2.x SRAM banks power
  ...

1 files changed, 332 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
new file mode 100644
index 000000000000..1c95ccc886c4
--- /dev/null
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -0,0 +1,332 @@
+/*
+ *  bytcr_rt5651.c - ASoc Machine driver for Intel Byt CR platform
+ *  (derived from bytcr_rt5640.c)
+ *
+ *  Copyright (C) 2015 Intel Corp
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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/acpi.h>
+#include <linux/device.h>
+#include <linux/dmi.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5651.h"
+#include "../atom/sst-atom-controls.h"
+
+static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Internal Mic", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
+	{"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"},
+
+	{"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */
+	{"IN2P", NULL, "Headset Mic"},
+	{"Headphone", NULL, "HPOL"},
+	{"Headphone", NULL, "HPOR"},
+	{"Speaker", NULL, "LOUTL"},
+	{"Speaker", NULL, "LOUTR"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic1_map[] = {
+	{"DMIC1", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic2_map[] = {
+	{"DMIC2", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = {
+	{"Internal Mic", NULL, "micbias1"},
+	{"IN1P", NULL, "Internal Mic"},
+};
+
+enum {
+	BYT_RT5651_DMIC1_MAP,
+	BYT_RT5651_DMIC2_MAP,
+	BYT_RT5651_IN1_MAP,
+};
+
+#define BYT_RT5651_MAP(quirk)	((quirk) & 0xff)
+#define BYT_RT5651_DMIC_EN	BIT(16)
+
+static unsigned long byt_rt5651_quirk = BYT_RT5651_DMIC1_MAP |
+					BYT_RT5651_DMIC_EN;
+
+static const struct snd_kcontrol_new byt_rt5651_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Internal Mic"),
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static int byt_rt5651_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, RT5651_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, RT5651_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 dmi_system_id byt_rt5651_quirk_table[] = {
+	{}
+};
+
+static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
+{
+	int ret;
+	struct snd_soc_card *card = runtime->card;
+	const struct snd_soc_dapm_route *custom_map;
+	int num_routes;
+
+	card->dapm.idle_bias_off = true;
+
+	dmi_check_system(byt_rt5651_quirk_table);
+	switch (BYT_RT5651_MAP(byt_rt5651_quirk)) {
+	case BYT_RT5651_IN1_MAP:
+		custom_map = byt_rt5651_intmic_in1_map;
+		num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map);
+		break;
+	case BYT_RT5651_DMIC2_MAP:
+		custom_map = byt_rt5651_intmic_dmic2_map;
+		num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic2_map);
+		break;
+	default:
+		custom_map = byt_rt5651_intmic_dmic1_map;
+		num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic1_map);
+	}
+
+	ret = snd_soc_add_card_controls(card, byt_rt5651_controls,
+					ARRAY_SIZE(byt_rt5651_controls));
+	if (ret) {
+		dev_err(card->dev, "unable to add card controls\n");
+		return ret;
+	}
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+
+	return ret;
+}
+
+static const struct snd_soc_pcm_stream byt_rt5651_dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	.rate_min = 48000,
+	.rate_max = 48000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static int byt_rt5651_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);
+	int ret;
+
+	/* 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);
+
+	/*
+	 * Default mode for SSP configuration is TDM 4 slot, override config
+	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
+	 * dai_set_tdm_slot() since there is no other API exposed
+	 */
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+				  SND_SOC_DAIFMT_I2S     |
+				  SND_SOC_DAIFMT_NB_IF   |
+				  SND_SOC_DAIFMT_CBS_CFS
+				  );
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
+		return ret;
+	}
+
+	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_rt5651_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_rt5651_aif1_ops = {
+	.startup = byt_rt5651_aif1_startup,
+};
+
+static struct snd_soc_ops byt_rt5651_be_ssp2_ops = {
+	.hw_params = byt_rt5651_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link byt_rt5651_dais[] = {
+	[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,
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &byt_rt5651_aif1_ops,
+	},
+	[MERR_DPCM_DEEP_BUFFER] = {
+		.name = "Deep-Buffer Audio Port",
+		.stream_name = "Deep-Buffer Audio",
+		.cpu_dai_name = "deepbuffer-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.ignore_suspend = 1,
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.ops = &byt_rt5651_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 = "rt5651-aif1",
+		.codec_name = "i2c-10EC5651:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = byt_rt5651_codec_fixup,
+		.ignore_suspend = 1,
+		.nonatomic = true,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.init = byt_rt5651_init,
+		.ops = &byt_rt5651_be_ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card byt_rt5651_card = {
+	.name = "bytcr-rt5651",
+	.owner = THIS_MODULE,
+	.dai_link = byt_rt5651_dais,
+	.num_links = ARRAY_SIZE(byt_rt5651_dais),
+	.dapm_widgets = byt_rt5651_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(byt_rt5651_widgets),
+	.dapm_routes = byt_rt5651_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map),
+	.fully_routed = true,
+};
+
+static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+
+	/* register the soc card */
+	byt_rt5651_card.dev = &pdev->dev;
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card);
+
+	if (ret_val) {
+		dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n",
+			ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &byt_rt5651_card);
+	return ret_val;
+}
+
+static struct platform_driver snd_byt_rt5651_mc_driver = {
+	.driver = {
+		.name = "bytcr_rt5651",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = snd_byt_rt5651_mc_probe,
+};
+
+module_platform_driver(snd_byt_rt5651_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver for RT5651");
+MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytcr_rt5651");