Merge tag 'sound-4.12-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It was a relatively calm development cycle, and no scaring changes are
  seen in both core and driver sides. Here are some highlights:

  ASoC:
   - A new API for hooking up jacks more generically and easily

   - Card longname is set based on DMI for a unique UCM profile

   - Lots of Intel driver fixes: Atom, Broxton, Skylake and newer chips

   - New drivers for Cirrus CS35L35, DIO DIO2125, Everest ES7132,
     HiSilicon hi6210, Maxim MAX98927, MT2701 systems with WM8960,
     Nuvoton NAU8824, Odroid systems, ST STM32 SAI controllers and x86
     systems with DA7213

  HD-audio:
   - Many new quirks to support headset for various devices (mostly ASUS
     ones) as usual

   - Support for dual codecs on some Gigabyte mobos and Lenovo laptop

   - Improvement on PCM position reporting for Skylake and newer

  FireWire:
   - New drivers for MOTU and RME Fireface series

   - Updates for Digidesign Digi00x and TASCAM series

   - Support for tracepoints

  Others:
   - USB-audio: improved support for quirk_alias option

   - Cleanups, constification allover the places"

* tag 'sound-4.12-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
  ASoC: codec: wm8960: Relax bit clock computation when using PLL
  ASoC: codec: wm9860: avoid maybe-uninitialized warning
  ASoC: nau8824: leave Class D gain at chip default
  ASoC: nau8824: rename controls to match DAPM controls
  ASoC: Intel: Skylake: Return negative error code
  ASoC: Intel: Skylake: Fix unused variable warning
  ASoC: Intel: Skylake: fix uninitialized pointer use
  ASoC: sti: Fix error handling if of_clk_get() fails
  ASoC: cs4271: configure reset GPIO as output
  ASoC: dwc: Disallow building designware_pcm as a module
  ALSA: ali5451: fix spelling mistake in "ali_capture_preapre"
  ASoC: stm32: add SAI driver
  ASoC: stm32: add bindings for SAI
  ASoC: Intel: Skylake: Add loadable module support on KBL platform
  ASoC: Intel: Skylake: Modify load_lib_ipc arguments for a nowait version
  ASoC: Intel: Skylake: Register dsp_fw_ops for kabylake
  ASoC: Intel: Skylake: Modify arguments to reuse module transfer function
  ASoC: Intel: Skylake: Commonize library load
  ASoC: Intel: Skylake: Move sst common initialization to a helper function
  ASoC: nau8824: new driver
  ...

1 files changed, 283 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c
new file mode 100644
index 000000000000..18873e23f404
--- /dev/null
+++ b/sound/soc/intel/boards/bytcht_da7213.c
@@ -0,0 +1,283 @@
+/*
+ *  bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and
+ *             Cherrytrail-based platforms, with Dialog DA7213 codec
+ *
+ *  Copyright (C) 2017 Intel Corporation
+ *  Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.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/module.h>
+#include <linux/acpi.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <asm/platform_sst_audio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/da7213.h"
+#include "../atom/sst-atom-controls.h"
+#include "../common/sst-acpi.h"
+
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Mic"),
+	SOC_DAPM_PIN_SWITCH("Aux In"),
+};
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPL"},
+	{"Headphone Jack", NULL, "HPR"},
+
+	{"AUXL", NULL, "Aux In"},
+	{"AUXR", NULL, "Aux In"},
+
+	/* Assume Mic1 is linked to Headset and Mic2 to on-board mic */
+	{"MIC1", NULL, "Headset Mic"},
+	{"MIC2", NULL, "Mic"},
+
+	/* SOC-codec link */
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx"},
+	{"codec_in1", NULL, "ssp2 Rx"},
+
+	{"Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Rx", NULL, "Capture"},
+};
+
+static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
+		       struct snd_pcm_hw_params *params)
+{
+	int ret;
+	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 convert 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_NF   |
+				  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 int aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, 48000);
+}
+
+static int 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;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
+				     19200000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0,
+			DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
+		return -EIO;
+	}
+
+	return ret;
+}
+
+static int aif1_hw_free(struct snd_pcm_substream *substream)
+{
+	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_pll(codec_dai, 0,
+				  DA7213_SYSCLK_MCLK, 0, 0);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
+		return -EIO;
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_ops aif1_ops = {
+	.startup = aif1_startup,
+};
+
+static const struct snd_soc_ops ssp2_ops = {
+	.hw_params = aif1_hw_params,
+	.hw_free = aif1_hw_free,
+
+};
+
+static struct snd_soc_dai_link 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",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &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",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.ops = &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",
+		.id = 1,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "da7213-hifi",
+		.codec_name = "i2c-DLGS7213:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = codec_fixup,
+		.nonatomic = true,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card bytcht_da7213_card = {
+	.name = "bytcht-da7213",
+	.owner = THIS_MODULE,
+	.dai_link = dailink,
+	.num_links = ARRAY_SIZE(dailink),
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
+	.dapm_widgets = dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */
+
+static int bytcht_da7213_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+	int i;
+	struct snd_soc_card *card;
+	struct sst_acpi_mach *mach;
+	const char *i2c_name = NULL;
+	int dai_index = 0;
+
+	mach = (&pdev->dev)->platform_data;
+	card = &bytcht_da7213_card;
+	card->dev = &pdev->dev;
+
+	/* fix index of codec dai */
+	dai_index = MERR_DPCM_COMPR + 1;
+	for (i = 0; i < ARRAY_SIZE(dailink); i++) {
+		if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) {
+			dai_index = i;
+			break;
+		}
+	}
+
+	/* fixup codec name based on HID */
+	i2c_name = sst_acpi_find_name_from_hid(mach->id);
+	if (i2c_name != NULL) {
+		snprintf(codec_name, sizeof(codec_name),
+			"%s%s", "i2c-", i2c_name);
+		dailink[dai_index].codec_name = codec_name;
+	}
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, card);
+	return ret_val;
+}
+
+static struct platform_driver bytcht_da7213_driver = {
+	.driver = {
+		.name = "bytcht_da7213",
+	},
+	.probe = bytcht_da7213_probe,
+};
+module_platform_driver(bytcht_da7213_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver");
+MODULE_AUTHOR("Pierre-Louis Bossart");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytcht_da7213");