Merge tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This looks like a relatively calm development cycle; there have been
  only few changes in ALSA and ASoC core sides while we get lots of
  device-specific fixes and updates as usual. Most of commits are about
  ASoC, including Intel SOF/AVS and many device tree updates.

  Below are some highlights:

  Core:
   - Improvement in memalloc helper for fallback allocations
   - More cleanups of ASoC DAPM code

  ASoC:
   - Factoring out of mapping hw_params onto SoundWire configuration
   - The ever ongoing overhauls of the Intel DSP code continue,
     including support for loading libraries and probes with IPC4 on
     SOF.
   - Support for more sample formats on JZ4740
   - Lots of device tree conversions and fixups
   - Support for Allwinner D1, a range of AMD and Intel systems,
     Mediatek systems with multiple DMICs, Nuvoton NAU8318, NXP
     fsl_rpmsg and i.MX93, Qualcomm AudioReach Enable, MFC and SAL,
     RealTek RT1318 and Rockchip RK3588

  ALSA:
   - Addition of PCM kselftest; still minimalistic but can be extended
     in future
   - Fixes for corner-case XRUNs with USB-audio implicit feedback mode
   - Usual device-specific quirk updates for USB- and HD-audio
   - FireWire DICE updates

  This also contains a few cross-tree updates:
   - Some OMAP board file updates for removal of relevant OMAP platforms
   - A new I2C API update for I2C probe API adaption
   - A DRM update for the further hdmi-codec updates"

* tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (417 commits)
  ALSA: mts64: fix possible null-ptr-defer in snd_mts64_interrupt
  ALSA: patch_realtek: Fix Dell Inspiron Plus 16
  ALSA: hda/cirrus: Add extra 10 ms delay to allow PLL settle and lock.
  ASoC: dt-bindings: Correct Alexandre Belloni email
  ASoC: dt-bindings: maxim,max98504: Convert to DT schema
  ASoC: dt-bindings: maxim,max98357a: Convert to DT schema
  ASoC: dt-bindings: Reference common DAI properties
  ASoC: dt-bindings: Extend name-prefix.yaml into common DAI properties
  ASoC: rt715: Make read-only arrays capture_reg_H and capture_reg_L static const
  ASoC: uniphier: aio-core: Make some read-only arrays static const
  ASoC: wcd938x: Make read-only array minCode_param static const
  ASoC: qcom: lpass-sc7280: Add maybe_unused tag for system PM ops
  ASoC : SOF: amd: Add support for IPC and DSP dumps
  ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER
  ALSA: usb-audio: Workaround for XRUN at prepare
  ALSA: pcm: Handle XRUN at trigger START
  ALSA: pcm: Set missing stop_operating flag at undoing trigger start
  drm: tda99x: Don't advertise non-existent capture support
  ASoC: hdmi-codec: Allow playback and capture to be disabled
  kselftest/alsa: Add more coverage of sample rates and channel counts
  ...

1 files changed, 236 insertions, 0 deletions

diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c
new file mode 100644
index 000000000000..7cccce99f92e
--- /dev/null
+++ b/sound/soc/intel/avs/boards/max98927.c
@@ -0,0 +1,236 @@
+// 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 MAX98927_DEV0_NAME	"i2c-MX98927:00"
+#define MAX98927_DEV1_NAME	"i2c-MX98927:01"
+#define MAX98927_CODEC_NAME	"max98927-aif1"
+
+static struct snd_soc_codec_conf card_codec_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF(MAX98927_DEV0_NAME),
+		.name_prefix = "Right",
+	},
+	{
+		.dlc = COMP_CODEC_CONF(MAX98927_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_max98927_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 convert 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_max98927_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 = 0;
+	int i;
+
+	for_each_rtd_codec_dais(runtime, i, codec_dai) {
+		if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME))
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+		else if (!strcmp(codec_dai->component->name, MAX98927_DEV1_NAME))
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+
+		if (ret < 0) {
+			dev_err(runtime->dev, "hw_params for %s failed: %d\n",
+				codec_dai->component->name, ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_ops avs_max98927_ops = {
+	.hw_params = avs_max98927_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, MAX98927_DEV0_NAME);
+	dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98927_CODEC_NAME);
+	dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, MAX98927_DEV1_NAME);
+	dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98927_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_CBS_CFS;
+	dl->be_hw_params_fixup = avs_max98927_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_max98927_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_max98927_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_max98927";
+	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_max98927_driver = {
+	.probe = avs_max98927_probe,
+	.driver = {
+		.name = "avs_max98927",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+
+module_platform_driver(avs_max98927_driver)
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:avs_max98927");