Merge tag 'sound-6.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Lots of changes as usual, but the only significant stuff in ALSA core
  part is the MIDI 2.0 support, while ASoC core kept receiving the code
  refactoring. The majority of changes are seen rather in device
  drivers, and quite a few new drivers can be found there.

  Here we go, some highlights:

  ALSA and ASoC Core:
   - Support of MIDI 2.0 devices: rawmidi and sequencer API have been
     extended for the support of the new UMP (Universal MIDI Packet)
     protocol, USB audio driver got the USB MIDI 2.0 interface support
   - Continued refactoring around ASoC DAI links and the ordering of
     trigger callbacks
   - PCM ABI extension for better drain support

  ASoC Drivers:
   - Conversions of many drivers to use maple tree based caches
   - Everlasting improvement works on ASoC Intel drivers
   - Compressed audio support for Qualcomm
   - Support for AMD SoundWire, Analog Devices SSM3515, Google
     Chameleon, Ingenic X1000, Intel systems with various CODECs,
     Loongson platforms, Maxim MAX98388, Mediatek MT8188, Nuvoton
     NAU8825C, NXP platforms with NAU8822, Qualcomm WSA884x, StarFive
     JH7110, Texas Instruments TAS2781

  HD-audio:
   - Quirks for HP and ASUS machines
   - CS35L41 HD-audio codec fixes
   - Loongson HD-audio support

  Misc:
   - A new virtual PCM test driver for kselftests
   - Continued refactoring and improvements on the legacy emu10k1
     driver"

* tag 'sound-6.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (556 commits)
  ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on EliteBook
  ASoC: hdmi-codec: fix channel info for compressed formats
  ALSA: pcm: fix ELD constraints for (E)AC3, DTS(-HD) and MLP formats
  ASoC: core: Always store of_node when getting DAI link component
  ASoC: tas2781: Fix error code in tas2781_load_calibration()
  ASoC: amd: update pm_runtime enable sequence
  ALSA: ump: Export MIDI1 / UMP conversion helpers
  ASoC: tas2781: fix Kconfig dependencies
  ASoC: amd: acp: remove acp poweroff function
  ASoC: amd: acp: clear pdm dma interrupt mask
  ASoC: codecs: max98090: Allow dsp_a mode
  ASoC: qcom: common: add default jack dapm pins
  ASoC: loongson: fix address space confusion
  ASoC: dt-bindings: microchip,sama7g5-pdmc: Simplify "microchip,mic-pos" constraints
  ASoC: tegra: Remove stale comments in AHUB
  ASoC: tegra: Use normal system sleep for ASRC
  ALSA: hda/realtek: Add quirks for ROG ALLY CS35l41 audio
  ASoC: fsl-asoc-card: Allow passing the number of slots in use
  ASoC: codecs: wsa884x: Add WSA884x family of speakers
  ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers
  ...

1 files changed, 105 insertions, 13 deletions

diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index e5ff61c1e9d1..3019626b0592 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -889,11 +889,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 	dev_dbg(dev, "link %d, dais %d, ccnf %d\n",
 		li->link, dai_num, cnf_num);
 
-	/* dummy CPU/Codec */
-	priv->dummy.of_node	= NULL;
-	priv->dummy.dai_name	= "snd-soc-dummy-dai";
-	priv->dummy.name	= "snd-soc-dummy";
-
 	priv->dai_props		= dai_props;
 	priv->dai_link		= dai_link;
 	priv->dais		= dais;
@@ -908,7 +903,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 	for (i = 0; i < li->link; i++) {
 		if (li->num[i].cpus) {
 			/* Normal CPU */
-			dai_props[i].cpus	=
 			dai_link[i].cpus	= dlcs;
 			dai_props[i].num.cpus	=
 			dai_link[i].num_cpus	= li->num[i].cpus;
@@ -918,15 +912,13 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 			dais += li->num[i].cpus;
 		} else {
 			/* DPCM Be's CPU = dummy */
-			dai_props[i].cpus	=
-			dai_link[i].cpus	= &priv->dummy;
+			dai_link[i].cpus	= &asoc_dummy_dlc;
 			dai_props[i].num.cpus	=
 			dai_link[i].num_cpus	= 1;
 		}
 
 		if (li->num[i].codecs) {
 			/* Normal Codec */
-			dai_props[i].codecs	=
 			dai_link[i].codecs	= dlcs;
 			dai_props[i].num.codecs	=
 			dai_link[i].num_codecs	= li->num[i].codecs;
@@ -942,15 +934,13 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 			}
 		} else {
 			/* DPCM Fe's Codec = dummy */
-			dai_props[i].codecs	=
-			dai_link[i].codecs	= &priv->dummy;
+			dai_link[i].codecs	= &asoc_dummy_dlc;
 			dai_props[i].num.codecs	=
 			dai_link[i].num_codecs	= 1;
 		}
 
 		if (li->num[i].platforms) {
 			/* Have Platform */
-			dai_props[i].platforms		=
 			dai_link[i].platforms		= dlcs;
 			dai_props[i].num.platforms	=
 			dai_link[i].num_platforms	= li->num[i].platforms;
@@ -958,7 +948,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 			dlcs += li->num[i].platforms;
 		} else {
 			/* Doesn't have Platform */
-			dai_props[i].platforms		=
 			dai_link[i].platforms		= NULL;
 			dai_props[i].num.platforms	=
 			dai_link[i].num_platforms	= 0;
@@ -1024,6 +1013,109 @@ int asoc_graph_is_ports0(struct device_node *np)
 }
 EXPORT_SYMBOL_GPL(asoc_graph_is_ports0);
 
+static int graph_get_dai_id(struct device_node *ep)
+{
+	struct device_node *node;
+	struct device_node *endpoint;
+	struct of_endpoint info;
+	int i, id;
+	int ret;
+
+	/* use driver specified DAI ID if exist */
+	ret = snd_soc_get_dai_id(ep);
+	if (ret != -ENOTSUPP)
+		return ret;
+
+	/* use endpoint/port reg if exist */
+	ret = of_graph_parse_endpoint(ep, &info);
+	if (ret == 0) {
+		/*
+		 * Because it will count port/endpoint if it doesn't have "reg".
+		 * But, we can't judge whether it has "no reg", or "reg = <0>"
+		 * only of_graph_parse_endpoint().
+		 * We need to check "reg" property
+		 */
+		if (of_property_present(ep,   "reg"))
+			return info.id;
+
+		node = of_get_parent(ep);
+		ret = of_property_present(node, "reg");
+		of_node_put(node);
+		if (ret)
+			return info.port;
+	}
+	node = of_graph_get_port_parent(ep);
+
+	/*
+	 * Non HDMI sound case, counting port/endpoint on its DT
+	 * is enough. Let's count it.
+	 */
+	i = 0;
+	id = -1;
+	for_each_endpoint_of_node(node, endpoint) {
+		if (endpoint == ep)
+			id = i;
+		i++;
+	}
+
+	of_node_put(node);
+
+	if (id < 0)
+		return -ENODEV;
+
+	return id;
+}
+
+int asoc_graph_parse_dai(struct device_node *ep,
+			 struct snd_soc_dai_link_component *dlc,
+			 int *is_single_link)
+{
+	struct device_node *node;
+	struct of_phandle_args args = {};
+	int ret;
+
+	if (!ep)
+		return 0;
+
+	node = of_graph_get_port_parent(ep);
+
+	/* Get dai->name */
+	args.np		= node;
+	args.args[0]	= graph_get_dai_id(ep);
+	args.args_count	= (of_graph_get_endpoint_count(node) > 1);
+
+	/*
+	 * FIXME
+	 *
+	 * Here, dlc->dai_name is pointer to CPU/Codec DAI name.
+	 * If user unbinded CPU or Codec driver, but not for Sound Card,
+	 * dlc->dai_name is keeping unbinded CPU or Codec
+	 * driver's pointer.
+	 *
+	 * If user re-bind CPU or Codec driver again, ALSA SoC will try
+	 * to rebind Card via snd_soc_try_rebind_card(), but because of
+	 * above reason, it might can't bind Sound Card.
+	 * Because Sound Card is pointing to released dai_name pointer.
+	 *
+	 * To avoid this rebind Card issue,
+	 * 1) It needs to alloc memory to keep dai_name eventhough
+	 *    CPU or Codec driver was unbinded, or
+	 * 2) user need to rebind Sound Card everytime
+	 *    if he unbinded CPU or Codec.
+	 */
+	ret = snd_soc_get_dlc(&args, dlc);
+	if (ret < 0) {
+		of_node_put(node);
+		return ret;
+	}
+
+	if (is_single_link)
+		*is_single_link = of_graph_get_endpoint_count(node) == 1;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(asoc_graph_parse_dai);
+
 /* Module information */
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
 MODULE_DESCRIPTION("ALSA SoC Simple Card Utils");