Merge tag 'v5.9-rc2' into asoc-5.10

Linux 5.9-rc2

6 files changed, 57 insertions, 17 deletions

diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 09ddab5f5cae..9766f6af8743 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
 	INIT_LIST_HEAD(&bus->hlink_list);
 	init_waitqueue_head(&bus->rirb_wq);
 	bus->irq = -1;
+
+	/*
+	 * Default value of '8' is as per the HD audio specification (Rev 1.0a).
+	 * Following relation is used to derive STRIPE control value.
+	 *  For sample rate <= 48K:
+	 *   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
+	 *  For sample rate > 48K:
+	 *   { ((num_channels * bits_per_sample * rate/48000) /
+	 *	number of SDOs) >= 8 }
+	 */
+	bus->sdo_limit = 8;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_init);
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 011b17cc1efa..b98449fd92f3 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset)
 
 	bus->chip_init = true;
 
-	/*
-	 * Default value of '8' is as per the HD audio specification (Rev 1.0a).
-	 * Following relation is used to derive STRIPE control value.
-	 *  For sample rate <= 48K:
-	 *   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
-	 *  For sample rate > 48K:
-	 *   { ((num_channels * bits_per_sample * rate/48000) /
-	 *	number of SDOs) >= 8 }
-	 */
-	bus->sdo_limit = 8;
-
 	return true;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip);
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 5363d88cc4b9..2e5a5c5279e8 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu)
 }
 
 /*
- * This is apparently the standard way to initailise an MPU-401
+ * This is apparently the standard way to initialise an MPU-401
  */
 static inline void initialise_mpu401(const struct snd_mpu401 *mpu)
 {
@@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c)
 }
 
 /*
- * Tell the SoundScape to begin a DMA tranfer using the given channel.
+ * Tell the SoundScape to begin a DMA transfer using the given channel.
  * All locking issues are left to the caller.
  */
 static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
@@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu)
 }
 
 /*
- * Initialse an MPU-401 subdevice for MIDI support on the SoundScape.
+ * Initialise an MPU-401 subdevice for MIDI support on the SoundScape.
  */
 static int create_mpu401(struct snd_card *card, int devnum,
 			 unsigned long port, int irq)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7f9d35273734..a1fa983d2a94 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7694,6 +7694,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
 	SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
@@ -7955,6 +7957,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
 	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
 	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
+	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 6b0f3a8469ef..81e987eaf063 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2371,7 +2371,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 	int num_ins;
 	struct usb_mixer_elem_info *cval;
 	struct snd_kcontrol *kctl;
-	int i, err, nameid, type, len;
+	int i, err, nameid, type, len, val;
 	const struct procunit_info *info;
 	const struct procunit_value_info *valinfo;
 	const struct usbmix_name_map *map;
@@ -2474,6 +2474,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 			break;
 		}
 
+		err = get_cur_ctl_value(cval, cval->control << 8, &val);
+		if (err < 0) {
+			usb_mixer_elem_info_free(cval);
+			return -EINVAL;
+		}
+
 		kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
 		if (!kctl) {
 			usb_mixer_elem_info_free(cval);
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d79e3ddc5690..f4fb002e3ef4 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2680,6 +2680,10 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.data = (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			{
+				.ifnum = 0,
 				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
 				.data = &(const struct audioformat) {
 					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
@@ -2690,6 +2694,32 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
 					.endpoint = 0x01,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.datainterval = 1,
+					.maxpacksize = 0x024c,
+					.rates = SNDRV_PCM_RATE_44100 |
+						 SNDRV_PCM_RATE_48000,
+					.rate_min = 44100,
+					.rate_max = 48000,
+					.nr_rates = 2,
+					.rate_table = (unsigned int[]) {
+						44100, 48000
+					}
+				}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 2,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = 0,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.datainterval = 1,
+					.maxpacksize = 0x0126,
 					.rates = SNDRV_PCM_RATE_44100 |
 						 SNDRV_PCM_RATE_48000,
 					.rate_min = 44100,
@@ -3714,8 +3744,8 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
  * they pretend to be 96kHz mono as a workaround for stereo being broken
  * by that...
  *
- * They also have swapped L-R channels, but that's for userspace to deal
- * with.
+ * They also have an issue with initial stream alignment that causes the
+ * channels to be swapped and out of phase, which is dealt with in quirks.c.
  */
 {
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |