Merge 3.0.54

Change-Id: I68316b1f2af49f27e3462384592d3bba217302b6

8 files changed, 16 insertions, 7 deletions

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d694e9d..7b6f959 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -465,6 +465,7 @@ static int ad198x_build_pcms(struct hda_codec *codec)
 	if (spec->multiout.dig_out_nid) {
 		info++;
 		codec->num_pcms++;
+		codec->spdif_status_reset = 1;
 		info->name = "AD198x Digital";
 		info->pcm_type = HDA_PCM_TYPE_SPDIF;
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index fb6fbe4..15bfe9d 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -991,7 +991,7 @@ static const struct hda_verb cs_coef_init_verbs[] = {
 	  | 0x0400 /* Disable Coefficient Auto increment */
 	  )},
 	/* Beep */
-	{0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+	{0x11, AC_VERB_SET_COEF_INDEX, IDX_BEEP_CFG},
 	{0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */
 
 	{} /* terminator */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8d288a7..3315816 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5967,6 +5967,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, const hda_nid_t *nids,
 	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
 
 static const struct snd_pci_quirk beep_white_list[] = {
+	SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1),
 	SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
 	SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
 	SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
@@ -20134,6 +20135,9 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
 	{ .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
 	{ .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
+	{ .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
+	{ .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
+	{ .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index ce33be0e..66488a7 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -223,7 +223,7 @@ static int pdacf_config(struct pcmcia_device *link)
 	if (ret)
 		goto failed;
 
-	ret = pcmcia_request_exclusive_irq(link, pdacf_interrupt);
+	ret = pcmcia_request_irq(link, pdacf_interrupt);
 	if (ret)
 		goto failed;
 
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index d9ef21d..31777d1 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -229,7 +229,7 @@ static int vxpocket_config(struct pcmcia_device *link)
 	if (ret)
 		goto failed;
 
-	ret = pcmcia_request_exclusive_irq(link, snd_vx_irq_handler);
+	ret = pcmcia_request_irq(link, snd_vx_irq_handler);
 	if (ret)
 		goto failed;
 
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 85e3e63..e0368d5 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -750,7 +750,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream,
 		wm8978->mclk_idx = -1;
 		f_sel = wm8978->f_mclk;
 	} else {
-		if (!wm8978->f_pllout) {
+		if (!wm8978->f_opclk) {
 			/* We only enter here, if OPCLK is not used */
 			int ret = wm8978_configure_pll(codec);
 			if (ret < 0)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d5ec206..0c9dee2 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2634,7 +2634,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
 {
 	struct snd_soc_codec *codec;
 
-	list_for_each_entry(codec, &card->codec_dev_list, list) {
+	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
 		soc_dapm_shutdown_codec(&codec->dapm);
 		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
 			snd_soc_dapm_set_bias_level(&codec->dapm,
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index f928910..a0fe5dd 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -147,6 +147,7 @@ struct snd_usb_midi_out_endpoint {
 		struct snd_usb_midi_out_endpoint* ep;
 		struct snd_rawmidi_substream *substream;
 		int active;
+		bool autopm_reference;
 		uint8_t cable;		/* cable number << 4 */
 		uint8_t state;
 #define STATE_UNKNOWN	0
@@ -1059,7 +1060,8 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream)
 		return -ENXIO;
 	}
 	err = usb_autopm_get_interface(umidi->iface);
-	if (err < 0)
+	port->autopm_reference = err >= 0;
+	if (err < 0 && err != -EACCES)
 		return -EIO;
 	substream->runtime->private_data = port;
 	port->state = STATE_UNKNOWN;
@@ -1070,9 +1072,11 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream)
 static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream)
 {
 	struct snd_usb_midi* umidi = substream->rmidi->private_data;
+	struct usbmidi_out_port *port = substream->runtime->private_data;
 
 	substream_open(substream, 0);
-	usb_autopm_put_interface(umidi->iface);
+	if (port->autopm_reference)
+		usb_autopm_put_interface(umidi->iface);
 	return 0;
 }