Merge tag 'sound-6.0-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "As the diffstat shows, we've had lots of developments in a wide range
  at this time; the majority of changes are about ASoC, including
  subsystem-wide cleanups, continued SOF / Intel updates and a bunch of
  new drivers (as usual), while there have been some significant (but
  almost invisible) improvements in ALSA core side, too.

  Below are some highlights:

  Core:

   - Faster lookups of control elements with Xarray; normal user won't
     notice, but on the devices with tons of control elements, it can be
     visibly faster

   - Support for input validation for controls; this will harden for
     badly written drivers in general with a slight overhead

   - Deferred async signal handling for working around the potential
     deadlocks

   - Cleanup / refactoring raw MIDI locking code

  ASoC:

   - Restructing of the set_fmt() callbacks for making things clearer in
     situations like CODEC to CODEC links

   - Clean up and modernizing the DAI naming scheme setups

   - Merge of more of the Intel AVS driver stack, including some board
     integrations

   - New version 4 mechanism for communication with SOF DSPs

   - Suppoort for dynamically selecting the PLL to use at runtime on
     i.MX platforms

   - Improvements for CODEC to CODEC support in the generic cards

   - Support for AMD Jadeite and various machines, AMD RPL, Intel
     MetorLake DSPs, Mediatek MT8186 DSPs and MT6366, nVidia Tegra
     MDDRC, OPE and PEQ, NXP TFA9890, Qualcomm SDM845, WCD9335 and
     WAS883x, and Texas Instruments TAS2780

  HD- and USB-audio:

   - Continued improvement for CS35L41 (sub)codec support

   - More quirks for various devices (HP, Lenovo, Dell, Clevo)"

* tag 'sound-6.0-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (778 commits)
  ALSA: hda/realtek: Add quirk for HP Spectre x360 15-eb0xxx
  ALSA: line6: Replace sprintf() with sysfs_emit()
  ALSA: hda: Replace sprintf() with sysfs_emit()
  ALSA: pcm: Replace sprintf() with sysfs_emit()
  ALSA: core: Replace scnprintf() with sysfs_emit()
  ALSA: control-led: Replace sprintf() with sysfs_emit()
  ALSA: aoa: Replace sprintf() with sysfs_emit()
  ALSA: ac97: Replace sprintf() with sysfs_emit()
  ALSA: hda/realtek: Add quirk for Clevo NV45PZ
  ALSA: hda/realtek: Add quirk for Lenovo Yoga9 14IAP7
  ALSA: control: Use deferred fasync helper
  ALSA: pcm: Use deferred fasync helper
  ALSA: timer: Use deferred fasync helper
  ALSA: core: Add async signal helpers
  ASoC: q6asm: use kcalloc() instead of kzalloc()
  ACPI: scan: Add CLSA0101 Laptop Support
  ALSA: hda: cs35l41: Support CLSA0101
  ALSA: hda: cs35l41: Use the CS35L41 HDA internal define
  ASoC: dt-bindings: use spi-peripheral-props.yaml
  ASoC: codecs: va-macro: use fsgen as clock
  ...

1 files changed, 239 insertions, 0 deletions

diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c
new file mode 100644
index 000000000000..0fa8f5606385
--- /dev/null
+++ b/sound/soc/intel/avs/boards/max98373.c
@@ -0,0 +1,239 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// Copyright(c) 2022 Intel Corporation. All rights reserved.
+//
+// Authors: Cezary Rojewski <cezary.rojewski@intel.com>
+//          Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
+//
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/soc-dapm.h>
+
+#define MAX98373_DEV0_NAME	"i2c-MX98373:00"
+#define MAX98373_DEV1_NAME	"i2c-MX98373:01"
+#define MAX98373_CODEC_NAME	"max98373-aif1"
+
+static struct snd_soc_codec_conf card_codec_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME),
+		.name_prefix = "Right",
+	},
+	{
+		.dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME),
+		.name_prefix = "Left",
+	},
+};
+
+static const struct snd_kcontrol_new card_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const struct snd_soc_dapm_widget card_widgets[] = {
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route card_base_routes[] = {
+	{ "Left Spk", NULL, "Left BE_OUT" },
+	{ "Right Spk", NULL, "Right BE_OUT" },
+};
+
+static int
+avs_max98373_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* 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 */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+static int avs_max98373_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai;
+	int ret, i;
+
+	for_each_rtd_codec_dais(runtime, i, codec_dai) {
+		if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+		if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_ops avs_max98373_ops = {
+	.hw_params = avs_max98373_hw_params,
+};
+
+static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
+			       struct snd_soc_dai_link **dai_link)
+{
+	struct snd_soc_dai_link_component *platform;
+	struct snd_soc_dai_link *dl;
+
+	dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL);
+	platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL);
+	if (!dl || !platform)
+		return -ENOMEM;
+
+	platform->name = platform_name;
+
+	dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port);
+	dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL);
+	dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs) * 2, GFP_KERNEL);
+	if (!dl->name || !dl->cpus || !dl->codecs)
+		return -ENOMEM;
+
+	dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port);
+	dl->codecs[0].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV0_NAME);
+	dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME);
+	dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV1_NAME);
+	dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME);
+	if (!dl->cpus->dai_name || !dl->codecs[0].name || !dl->codecs[0].dai_name ||
+	    !dl->codecs[1].name || !dl->codecs[1].dai_name)
+		return -ENOMEM;
+
+	dl->num_cpus = 1;
+	dl->num_codecs = 2;
+	dl->platforms = platform;
+	dl->num_platforms = 1;
+	dl->id = 0;
+	dl->dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC;
+	dl->be_hw_params_fixup = avs_max98373_be_fixup;
+	dl->nonatomic = 1;
+	dl->no_pcm = 1;
+	dl->dpcm_capture = 1;
+	dl->dpcm_playback = 1;
+	dl->ignore_pmdown_time = 1;
+	dl->ops = &avs_max98373_ops;
+
+	*dai_link = dl;
+
+	return 0;
+}
+
+static int avs_create_dapm_routes(struct device *dev, int ssp_port,
+				  struct snd_soc_dapm_route **routes, int *num_routes)
+{
+	struct snd_soc_dapm_route *dr;
+	const int num_base = ARRAY_SIZE(card_base_routes);
+	const int num_dr = num_base + 2;
+	int idx;
+
+	dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL);
+	if (!dr)
+		return -ENOMEM;
+
+	memcpy(dr, card_base_routes, num_base * sizeof(*dr));
+
+	idx = num_base;
+	dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Left HiFi Playback");
+	dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port);
+	if (!dr[idx].sink || !dr[idx].source)
+		return -ENOMEM;
+
+	idx++;
+	dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Right HiFi Playback");
+	dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port);
+	if (!dr[idx].sink || !dr[idx].source)
+		return -ENOMEM;
+
+	*routes = dr;
+	*num_routes = num_dr;
+
+	return 0;
+}
+
+static int avs_max98373_probe(struct platform_device *pdev)
+{
+	struct snd_soc_dapm_route *routes;
+	struct snd_soc_dai_link *dai_link;
+	struct snd_soc_acpi_mach *mach;
+	struct snd_soc_card *card;
+	struct device *dev = &pdev->dev;
+	const char *pname;
+	int num_routes, ssp_port, ret;
+
+	mach = dev_get_platdata(dev);
+	pname = mach->mach_params.platform;
+	ssp_port = __ffs(mach->mach_params.i2s_link_mask);
+
+	ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link);
+	if (ret) {
+		dev_err(dev, "Failed to create dai link: %d", ret);
+		return ret;
+	}
+
+	ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes);
+	if (ret) {
+		dev_err(dev, "Failed to create dapm routes: %d", ret);
+		return ret;
+	}
+
+	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+	if (!card)
+		return -ENOMEM;
+
+	card->name = "avs_max98373";
+	card->dev = dev;
+	card->owner = THIS_MODULE;
+	card->dai_link = dai_link;
+	card->num_links = 1;
+	card->codec_conf = card_codec_conf;
+	card->num_configs = ARRAY_SIZE(card_codec_conf);
+	card->controls = card_controls;
+	card->num_controls = ARRAY_SIZE(card_controls);
+	card->dapm_widgets = card_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(card_widgets);
+	card->dapm_routes = routes;
+	card->num_dapm_routes = num_routes;
+	card->fully_routed = true;
+
+	ret = snd_soc_fixup_dai_links_platform_name(card, pname);
+	if (ret)
+		return ret;
+
+	return devm_snd_soc_register_card(dev, card);
+}
+
+static struct platform_driver avs_max98373_driver = {
+	.probe = avs_max98373_probe,
+	.driver = {
+		.name = "avs_max98373",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+
+module_platform_driver(avs_max98373_driver)
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:avs_max98373");