Merge remote-tracking branch 'takashi/for-next' into sound/upstream-20260402

Author: bardliao

Documentation/sound/alsa-configuration.rst

@@ -2376,6 +2376,13 @@ quirk_flags
           Skip the probe-time interface setup (usb_set_interface,
           init_pitch, init_sample_rate); redundant with
           snd_usb_endpoint_prepare() at stream-open time
+        * bit 27: ``mixer_playback_linear_vol``
+          Set linear volume mapping for devices where the playback volume
+          control value is mapped to voltage (instead of dB) level linearly.
+          In short: ``x(raw) = (raw - raw_min) / (raw_max - raw_min)``;
+          ``V(x) = k * x``; ``dB(x) = 20 * log10(x)``. Overrides bit 24
+        * bit 28: ``mixer_capture_linear_vol``
+          Similar to bit 27 but for capture streams. Overrides bit 25
 
 This module supports multiple devices, autoprobe and hotplugging.
 

include/sound/core.h

@@ -133,6 +133,9 @@ struct snd_card {
 #ifdef CONFIG_SND_DEBUG
 	struct dentry *debugfs_root;    /* debugfs root for card */
 #endif
+#ifdef CONFIG_SND_CTL_DEBUG
+	struct snd_ctl_elem_value *value_buf; /* buffer for kctl->put() verification */
+#endif
 
 #ifdef CONFIG_PM
 	unsigned int power_state;	/* power state */

include/sound/hda_verbs.h

@@ -56,7 +56,12 @@ enum {
 #define AC_VERB_GET_DIGI_CONVERT_1		0x0f0d
 #define AC_VERB_GET_DIGI_CONVERT_2		0x0f0e /* unused */
 #define AC_VERB_GET_VOLUME_KNOB_CONTROL		0x0f0f
-/* f10-f1a: GPIO */
+/* f10-f1a: GPI/GPO/GPIO */
+#define AC_VERB_GET_GPI_DATA			0x0f10
+#define AC_VERB_GET_GPI_WAKE_MASK		0x0f11
+#define AC_VERB_GET_GPI_UNSOLICITED_RSP_MASK	0x0f12
+#define AC_VERB_GET_GPI_STICKY_MASK		0x0f13
+#define AC_VERB_GET_GPO_DATA			0x0f14
 #define AC_VERB_GET_GPIO_DATA			0x0f15
 #define AC_VERB_GET_GPIO_MASK			0x0f16
 #define AC_VERB_GET_GPIO_DIRECTION		0x0f17
@@ -99,6 +104,11 @@ enum {
 #define AC_VERB_SET_DIGI_CONVERT_2		0x70e
 #define AC_VERB_SET_DIGI_CONVERT_3		0x73e
 #define AC_VERB_SET_VOLUME_KNOB_CONTROL		0x70f
+#define AC_VERB_SET_GPI_DATA			0x710
+#define AC_VERB_SET_GPI_WAKE_MASK		0x711
+#define AC_VERB_SET_SPI_UNSOLICITED_RSP_MASK	0x712
+#define AC_VERB_SET_GPI_STICKY_MASK		0x713
+#define AC_VERB_SET_GPO_DATA			0x714
 #define AC_VERB_SET_GPIO_DATA			0x715
 #define AC_VERB_SET_GPIO_MASK			0x716
 #define AC_VERB_SET_GPIO_DIRECTION		0x717

include/sound/pcm.h

@@ -729,6 +729,10 @@ static inline void __snd_pcm_set_state(struct snd_pcm_runtime *runtime,
 	runtime->status->state = state; /* copy for mmap */
 }
 
+void snd_pcm_set_state(struct snd_pcm_substream *substream,
+		       snd_pcm_state_t state);
+snd_pcm_state_t snd_pcm_get_state(struct snd_pcm_substream *substream);
+
 /**
  * bytes_to_samples - Unit conversion of the size from bytes to samples
  * @runtime: PCM runtime instance

include/sound/timer.h

@@ -102,6 +102,7 @@ struct snd_timer_instance {
 	unsigned int slave_id;
 	struct list_head open_list;
 	struct list_head active_list;
+	struct list_head master_list;
 	struct list_head ack_list;
 	struct list_head slave_list_head;
 	struct list_head slave_active_head;

sound/aoa/aoa.h

@@ -48,7 +48,7 @@ struct aoa_codec {
 	u32 connected;
 
 	/* data the fabric can associate with this structure */
-	void *fabric_data;
+	const void *fabric_data;
 
 	/* private! */
 	struct list_head list;

sound/aoa/fabrics/layout.c

@@ -55,7 +55,7 @@ struct codec_connection {
 
 struct codec_connect_info {
 	char *name;
-	struct codec_connection *connections;
+	const struct codec_connection *connections;
 };
 
 #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF	(1<<0)
@@ -116,7 +116,7 @@ MODULE_ALIAS("aoa-device-id-35");
 MODULE_ALIAS("aoa-device-id-44");
 
 /* onyx with all but microphone connected */
-static struct codec_connection onyx_connections_nomic[] = {
+static const struct codec_connection onyx_connections_nomic[] = {
 	{
 		.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
 		.codec_bit = 0,
@@ -133,7 +133,7 @@ static struct codec_connection onyx_connections_nomic[] = {
 };
 
 /* onyx on machines without headphone */
-static struct codec_connection onyx_connections_noheadphones[] = {
+static const struct codec_connection onyx_connections_noheadphones[] = {
 	{
 		.connected = CC_SPEAKERS | CC_LINEOUT |
 			     CC_LINEOUT_LABELLED_HEADPHONE,
@@ -157,7 +157,7 @@ static struct codec_connection onyx_connections_noheadphones[] = {
 };
 
 /* onyx on machines with real line-out */
-static struct codec_connection onyx_connections_reallineout[] = {
+static const struct codec_connection onyx_connections_reallineout[] = {
 	{
 		.connected = CC_SPEAKERS | CC_LINEOUT | CC_HEADPHONE,
 		.codec_bit = 0,
@@ -174,7 +174,7 @@ static struct codec_connection onyx_connections_reallineout[] = {
 };
 
 /* tas on machines without line out */
-static struct codec_connection tas_connections_nolineout[] = {
+static const struct codec_connection tas_connections_nolineout[] = {
 	{
 		.connected = CC_SPEAKERS | CC_HEADPHONE,
 		.codec_bit = 0,
@@ -191,7 +191,7 @@ static struct codec_connection tas_connections_nolineout[] = {
 };
 
 /* tas on machines with neither line out nor line in */
-static struct codec_connection tas_connections_noline[] = {
+static const struct codec_connection tas_connections_noline[] = {
 	{
 		.connected = CC_SPEAKERS | CC_HEADPHONE,
 		.codec_bit = 0,
@@ -204,7 +204,7 @@ static struct codec_connection tas_connections_noline[] = {
 };
 
 /* tas on machines without microphone */
-static struct codec_connection tas_connections_nomic[] = {
+static const struct codec_connection tas_connections_nomic[] = {
 	{
 		.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
 		.codec_bit = 0,
@@ -217,7 +217,7 @@ static struct codec_connection tas_connections_nomic[] = {
 };
 
 /* tas on machines with everything connected */
-static struct codec_connection tas_connections_all[] = {
+static const struct codec_connection tas_connections_all[] = {
 	{
 		.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
 		.codec_bit = 0,
@@ -233,31 +233,31 @@ static struct codec_connection tas_connections_all[] = {
 	{} /* terminate array by .connected == 0 */
 };
 
-static struct codec_connection toonie_connections[] = {
+static const struct codec_connection toonie_connections[] = {
 	{
 		.connected = CC_SPEAKERS | CC_HEADPHONE,
 		.codec_bit = 0,
 	},
 	{} /* terminate array by .connected == 0 */
 };
 
-static struct codec_connection topaz_input[] = {
+static const struct codec_connection topaz_input[] = {
 	{
 		.connected = CC_DIGITALIN,
 		.codec_bit = 0,
 	},
 	{} /* terminate array by .connected == 0 */
 };
 
-static struct codec_connection topaz_output[] = {
+static const struct codec_connection topaz_output[] = {
 	{
 		.connected = CC_DIGITALOUT,
 		.codec_bit = 1,
 	},
 	{} /* terminate array by .connected == 0 */
 };
 
-static struct codec_connection topaz_inout[] = {
+static const struct codec_connection topaz_inout[] = {
 	{
 		.connected = CC_DIGITALIN,
 		.codec_bit = 0,
@@ -772,7 +772,7 @@ static int check_codec(struct aoa_codec *codec,
 {
 	const u32 *ref;
 	char propname[32];
-	struct codec_connection *cc;
+	const struct codec_connection *cc;
 
 	/* if the codec has a 'codec' node, we require a reference */
 	if (of_node_name_eq(codec->node, "codec")) {
@@ -895,7 +895,7 @@ static void layout_notify(void *data)
 
 static void layout_attached_codec(struct aoa_codec *codec)
 {
-	struct codec_connection *cc;
+	const struct codec_connection *cc;
 	struct snd_kcontrol *ctl;
 	int headphones, lineout;
 	struct layout_dev *ldev = layout_device;

sound/aoa/soundbus/i2sbus/core.c

@@ -84,6 +84,7 @@ static void i2sbus_release_dev(struct device *dev)
 	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++)
 		free_irq(i2sdev->interrupts[i], i2sdev);
 	i2sbus_control_remove_dev(i2sdev->control, i2sdev);
+	of_node_put(i2sdev->sound.ofdev.dev.of_node);
 	mutex_destroy(&i2sdev->lock);
 	kfree(i2sdev);
 }
@@ -147,7 +148,6 @@ static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index,
 }
 
 /* Returns 1 if added, 0 for otherwise; don't return a negative value! */
-/* FIXME: look at device node refcounting */
 static int i2sbus_add_dev(struct macio_dev *macio,
 			  struct i2sbus_control *control,
 			  struct device_node *np)
@@ -178,8 +178,9 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	i = 0;
 	for_each_child_of_node(np, child) {
 		if (of_node_name_eq(child, "sound")) {
+			of_node_put(sound);
 			i++;
-			sound = child;
+			sound = of_node_get(child);
 		}
 	}
 	if (i == 1) {
@@ -205,6 +206,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 			}
 		}
 	}
+	of_node_put(sound);
 	/* for the time being, until we can handle non-layout-id
 	 * things in some fabric, refuse to attach if there is no
 	 * layout-id property or we haven't been forced to attach.
@@ -219,7 +221,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	mutex_init(&dev->lock);
 	spin_lock_init(&dev->low_lock);
 	dev->sound.ofdev.archdata.dma_mask = macio->ofdev.archdata.dma_mask;
-	dev->sound.ofdev.dev.of_node = np;
+	dev->sound.ofdev.dev.of_node = of_node_get(np);
 	dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.archdata.dma_mask;
 	dev->sound.ofdev.dev.parent = &macio->ofdev.dev;
 	dev->sound.ofdev.dev.release = i2sbus_release_dev;
@@ -327,6 +329,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	for (i=0;i<3;i++)
 		release_and_free_resource(dev->allocated_resource[i]);
 	mutex_destroy(&dev->lock);
+	of_node_put(dev->sound.ofdev.dev.of_node);
 	kfree(dev);
 	return 0;
 }
@@ -405,6 +408,9 @@ static int i2sbus_resume(struct macio_dev* dev)
 	int err, ret = 0;
 
 	list_for_each_entry(i2sdev, &control->list, item) {
+		if (list_empty(&i2sdev->sound.codec_list))
+			continue;
+
 		/* reset i2s bus format etc. */
 		i2sbus_pcm_prepare_both(i2sdev);