Merge branch 'master' of git://git.kernel.org/pub/scm/linux/kernel/git/linville/wireless

Conflicts:
	net/mac80211/debugfs_sta.c
	net/mac80211/sta_info.h

27 files changed, 196 insertions, 86 deletions

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 6fd9391..4fa1dbd 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -133,7 +133,7 @@ static int atmel_abdac_prepare_dma(struct atmel_abdac *dac,
 	period_len = frames_to_bytes(runtime, runtime->period_size);
 
 	cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len,
-			period_len, DMA_TO_DEVICE);
+			period_len, DMA_MEM_TO_DEV);
 	if (IS_ERR(cdesc)) {
 		dev_dbg(&dac->pdev->dev, "could not prepare cyclic DMA\n");
 		return PTR_ERR(cdesc);
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 73516f6..61dade6 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -102,7 +102,7 @@ static void atmel_ac97c_dma_capture_period_done(void *arg)
 
 static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip,
 		struct snd_pcm_substream *substream,
-		enum dma_data_direction direction)
+		enum dma_transfer_direction direction)
 {
 	struct dma_chan			*chan;
 	struct dw_cyclic_desc		*cdesc;
@@ -118,7 +118,7 @@ static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip,
 		return -EINVAL;
 	}
 
-	if (direction == DMA_TO_DEVICE)
+	if (direction == DMA_MEM_TO_DEV)
 		chan = chip->dma.tx_chan;
 	else
 		chan = chip->dma.rx_chan;
@@ -133,7 +133,7 @@ static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip,
 		return PTR_ERR(cdesc);
 	}
 
-	if (direction == DMA_TO_DEVICE) {
+	if (direction == DMA_MEM_TO_DEV) {
 		cdesc->period_callback = atmel_ac97c_dma_playback_period_done;
 		set_bit(DMA_TX_READY, &chip->flags);
 	} else {
@@ -393,7 +393,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
 	if (cpu_is_at32ap7000()) {
 		if (!test_bit(DMA_TX_READY, &chip->flags))
 			retval = atmel_ac97c_prepare_dma(chip, substream,
-					DMA_TO_DEVICE);
+					DMA_MEM_TO_DEV);
 	} else {
 		/* Initialize and start the PDC */
 		writel(runtime->dma_addr, chip->regs + ATMEL_PDC_TPR);
@@ -484,7 +484,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
 	if (cpu_is_at32ap7000()) {
 		if (!test_bit(DMA_RX_READY, &chip->flags))
 			retval = atmel_ac97c_prepare_dma(chip, substream,
-					DMA_FROM_DEVICE);
+					DMA_DEV_TO_MEM);
 	} else {
 		/* Initialize and start the PDC */
 		writel(runtime->dma_addr, chip->regs + ATMEL_PDC_RPR);
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index ad40938..b413ed0 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -12,6 +12,9 @@ config SND_HWDEP
 config SND_RAWMIDI
 	tristate
 
+config SND_COMPRESS_OFFLOAD
+	tristate
+
 # To be effective this also requires INPUT - users should say:
 #    select SND_JACK if INPUT=y || INPUT=SND
 # to avoid having to force INPUT on.
@@ -154,16 +157,6 @@ config SND_DYNAMIC_MINORS
 
 	  If you are unsure about this, say N here.
 
-config SND_COMPRESS_OFFLOAD
-	tristate "ALSA Compressed audio offload support"
-	default n
-	help
-	  If you want support for offloading compressed audio and have such
-	  a hardware, then you should say Y here and also to the DSP driver
-	  of your platform.
-
-	  If you are unsure about this, say N here.
-
 config SND_SUPPORT_OLD_API
 	bool "Support old ALSA API"
 	default y
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 762bb10..f13ad53 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -268,8 +268,14 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		card->shortname, chip->io, chip->irq);
 
 	// (4) Alloc components.
+	err = snd_vortex_mixer(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
 	// ADB pcm.
-	if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) {
+	err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_PCM);
+	if (err < 0) {
 		snd_card_free(card);
 		return err;
 	}
@@ -299,11 +305,6 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return err;
 	}
 #endif
-	// snd_ac97_mixer and Vortex mixer.
-	if ((err = snd_vortex_mixer(chip)) < 0) {
-		snd_card_free(card);
-		return err;
-	}
 	if ((err = snd_vortex_midi(chip)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index 02f6e08..bb93815 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -105,6 +105,7 @@
 #define MIX_SPDIF(x) (vortex->mixspdif[x])
 
 #define NR_WTPB 0x20		/* WT channels per each bank. */
+#define NR_PCM	0x10
 
 /* Structs */
 typedef struct {
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 0488633..0ef2f97 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -168,6 +168,7 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
 			runtime->hw = snd_vortex_playback_hw_adb;
 #ifdef CHIP_AU8830
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+			VORTEX_IS_QUAD(vortex) &&
 			VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) {
 			runtime->hw.channels_max = 4;
 			snd_pcm_hw_constraint_list(runtime, 0,
diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
index 5b68435..501501e 100644
--- a/sound/pci/hda/alc880_quirks.c
+++ b/sound/pci/hda/alc880_quirks.c
@@ -762,16 +762,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec,
 	/* Looks like the unsol event is incompatible with the standard
 	 * definition.  4bit tag is placed at 28 bit!
 	 */
-	switch (res >> 28) {
+	res >>= 28;
+	switch (res) {
 	case ALC_MIC_EVENT:
 		alc88x_simple_mic_automute(codec);
 		break;
 	default:
-		alc_sku_unsol_event(codec, res);
+		alc_exec_unsol_event(codec, res);
 		break;
 	}
 }
 
+static void alc880_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	alc_exec_unsol_event(codec, res >> 28);
+}
+
 static void alc880_uniwill_p53_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -800,10 +806,11 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
 	/* Looks like the unsol event is incompatible with the standard
 	 * definition.  4bit tag is placed at 28 bit!
 	 */
-	if ((res >> 28) == ALC_DCVOL_EVENT)
+	res >>= 28;
+	if (res == ALC_DCVOL_EVENT)
 		alc880_uniwill_p53_dcvol_automute(codec);
 	else
-		alc_sku_unsol_event(codec, res);
+		alc_exec_unsol_event(codec, res);
 }
 
 /*
@@ -1677,7 +1684,7 @@ static const struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_lg_ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc880_lg_capture_source,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc880_unsol_event,
 		.setup = alc880_lg_setup,
 		.init_hook = alc_hp_automute,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
index bdf0ed4..bb364a5 100644
--- a/sound/pci/hda/alc882_quirks.c
+++ b/sound/pci/hda/alc882_quirks.c
@@ -730,6 +730,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
 		alc889A_mb31_automute(codec);
 }
 
+static void alc882_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	alc_exec_unsol_event(codec, res >> 26);
+}
+
 /*
  * configuration and preset
  */
@@ -775,7 +780,7 @@ static const struct alc_config_preset alc882_presets[] = {
 			.channel_mode = alc885_mba21_ch_modes,
 			.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
 			.input_mux = &alc882_capture_source,
-			.unsol_event = alc_sku_unsol_event,
+			.unsol_event = alc882_unsol_event,
 			.setup = alc885_mba21_setup,
 			.init_hook = alc_hp_automute,
        },
@@ -791,7 +796,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &alc882_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_mbp3_setup,
 		.init_hook = alc_hp_automute,
 	},
@@ -806,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &mb5_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_mb5_setup,
 		.init_hook = alc_hp_automute,
 	},
@@ -821,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &macmini3_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_macmini3_setup,
 		.init_hook = alc_hp_automute,
 	},
@@ -836,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = {
 		.input_mux = &alc889A_imac91_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_sku_unsol_event,
+		.unsol_event = alc882_unsol_event,
 		.setup = alc885_imac91_setup,
 		.init_hook = alc_hp_automute,
 	},
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0852e20..95dfb68 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -469,6 +469,7 @@ struct azx {
 	unsigned int irq_pending_warned :1;
 	unsigned int probing :1; /* codec probing phase */
 	unsigned int snoop:1;
+	unsigned int align_buffer_size:1;
 
 	/* for debugging */
 	unsigned int last_cmd[AZX_MAX_CODECS];
@@ -1690,7 +1691,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	runtime->hw.rates = hinfo->rates;
 	snd_pcm_limit_hw_rates(runtime);
 	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
-	if (align_buffer_size)
+	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory
 		   access but isn't required by the HDA spec and
@@ -2498,6 +2499,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
@@ -2772,8 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	/* disable buffer size rounding to 128-byte multiples if supported */
+	chip->align_buffer_size = align_buffer_size;
 	if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
-		align_buffer_size = 0;
+		chip->align_buffer_size = 0;
 
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 8a32a69..a7a5733 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e82acf..c95c8bd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -185,7 +185,6 @@ struct alc_spec {
 	unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
 	unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
 	unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
-	unsigned int use_jack_tbl:1; /* 1 for model=auto */
 
 	/* auto-mute control */
 	int automute_mode;
@@ -621,17 +620,10 @@ static void alc_mic_automute(struct hda_codec *codec)
 		alc_mux_select(codec, 0, spec->int_mic_idx, false);
 }
 
-/* unsolicited event for HP jack sensing */
-static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+/* handle the specified unsol action (ALC_XXX_EVENT) */
+static void alc_exec_unsol_event(struct hda_codec *codec, int action)
 {
-	struct alc_spec *spec = codec->spec;
-	if (codec->vendor_id == 0x10ec0880)
-		res >>= 28;
-	else
-		res >>= 26;
-	if (spec->use_jack_tbl)
-		res = snd_hda_jack_get_action(codec, res);
-	switch (res) {
+	switch (action) {
 	case ALC_HP_EVENT:
 		alc_hp_automute(codec);
 		break;
@@ -645,6 +637,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
 	snd_hda_jack_report_sync(codec);
 }
 
+/* unsolicited event for HP jack sensing */
+static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	if (codec->vendor_id == 0x10ec0880)
+		res >>= 28;
+	else
+		res >>= 26;
+	res = snd_hda_jack_get_action(codec, res);
+	alc_exec_unsol_event(codec, res);
+}
+
 /* call init functions of standard auto-mute helpers */
 static void alc_inithook(struct hda_codec *codec)
 {
@@ -1883,7 +1886,7 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = {
 };
 #endif
 
-static int alc_build_controls(struct hda_codec *codec)
+static int __alc_build_controls(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	struct snd_kcontrol *kctl = NULL;
@@ -2029,11 +2032,16 @@ static int alc_build_controls(struct hda_codec *codec)
 
 	alc_free_kctls(codec); /* no longer needed */
 
-	err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
+	return 0;
+}
+
+static int alc_build_controls(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int err = __alc_build_controls(codec);
 	if (err < 0)
 		return err;
-
-	return 0;
+	return snd_hda_jack_add_kctls(codec, &spec->autocfg);
 }
 
 
@@ -3233,7 +3241,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
 	int i, err, noutputs;
 
 	noutputs = cfg->line_outs;
-	if (spec->multi_ios > 0)
+	if (spec->multi_ios > 0 && cfg->line_outs < 3)
 		noutputs += spec->multi_ios;
 
 	for (i = 0; i < noutputs; i++) {
@@ -3904,7 +3912,6 @@ static void set_capture_mixer(struct hda_codec *codec)
 static void alc_auto_init_std(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	spec->use_jack_tbl = 1;
 	alc_auto_init_multi_out(codec);
 	alc_auto_init_extra_out(codec);
 	alc_auto_init_analog_input(codec);
@@ -4168,6 +4175,8 @@ static int patch_alc880(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC_MODEL_AUTO)
 		spec->init_hook = alc_auto_init_std;
+	else
+		codec->patch_ops.build_controls = __alc_build_controls;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc880_loopbacks;
@@ -4297,6 +4306,8 @@ static int patch_alc260(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC_MODEL_AUTO)
 		spec->init_hook = alc_auto_init_std;
+	else
+		codec->patch_ops.build_controls = __alc_build_controls;
 	spec->shutup = alc_eapd_shutup;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
@@ -4691,6 +4702,8 @@ static int patch_alc882(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC_MODEL_AUTO)
 		spec->init_hook = alc_auto_init_std;
+	else
+		codec->patch_ops.build_controls = __alc_build_controls;
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 87e684f..336cfcd 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1596,7 +1596,7 @@ static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
 				"Dell Studio 1557", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe,
-				"Dell Studio XPS 1645", STAC_DELL_M6_BOTH),
+				"Dell Studio XPS 1645", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413,
 				"Dell Studio 1558", STAC_DELL_M6_DMIC),
 	{} /* terminator */
@@ -1608,7 +1608,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
-		      "Alienware M17x", STAC_ALIENWARE_M17X),
+		      "Alienware M17x R3", STAC_DELL_EQ),
 	{} /* terminator */
 };
 
@@ -4163,13 +4163,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
 	return 1;
 }
 
-static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
+static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
 {
 	int i;
 	for (i = 0; i < cfg->hp_outs; i++)
 		if (cfg->hp_pins[i] == nid)
 			return 1; /* nid is a HP-Out */
-
+	for (i = 0; i < cfg->line_outs; i++)
+		if (cfg->line_out_pins[i] == nid)
+			return 1; /* nid is a line-Out */
 	return 0; /* nid is not a HP-Out */
 };
 
@@ -4375,7 +4377,7 @@ static int stac92xx_init(struct hda_codec *codec)
 			continue;
 		}
 
-		if (is_nid_hp_pin(cfg, nid))
+		if (is_nid_out_jack_pin(cfg, nid))
 			continue; /* already has an unsol event */
 
 		pinctl = snd_hda_codec_read(codec, nid, 0,
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 40b181b..9f3b01b 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -95,13 +95,13 @@ module_param(ac97_quirk, charp, 0444);
 MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
 module_param(buggy_semaphore, bool, 0444);
 MODULE_PARM_DESC(buggy_semaphore, "Enable workaround for hardwares with problematic codec semaphores.");
-module_param(buggy_irq, bool, 0444);
+module_param(buggy_irq, bint, 0444);
 MODULE_PARM_DESC(buggy_irq, "Enable workaround for buggy interrupts on some motherboards.");
 module_param(xbox, bool, 0444);
 MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection.");
 module_param(spdif_aclink, int, 0444);
 MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link.");
-module_param(inside_vm, bool, 0444);
+module_param(inside_vm, bint, 0444);
 MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization.");
 
 /* just for backward compatibility */
diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c
index 478303e..63cff90 100644
--- a/sound/pci/oxygen/xonar_wm87x6.c
+++ b/sound/pci/oxygen/xonar_wm87x6.c
@@ -177,6 +177,7 @@ static void wm8776_registers_init(struct oxygen *chip)
 	struct xonar_wm87x6 *data = chip->model_data;
 
 	wm8776_write(chip, WM8776_RESET, 0);
+	wm8776_write(chip, WM8776_PHASESWAP, WM8776_PH_MASK);
 	wm8776_write(chip, WM8776_DACCTRL1, WM8776_DZCEN |
 		     WM8776_PL_LEFT_LEFT | WM8776_PL_RIGHT_RIGHT);
 	wm8776_write(chip, WM8776_DACMUTE, chip->dac_mute ? WM8776_DMUTE : 0);
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index e908a81..a561040 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -1,13 +1,13 @@
 ##
-## Au1200/Au1550 PSC + DBDMA
+## Au1200/Au1550/Au1300 PSC + DBDMA
 ##
 config SND_SOC_AU1XPSC
-	tristate "SoC Audio for Au1200/Au1250/Au1550"
+	tristate "SoC Audio for Au12xx/Au13xx/Au1550"
 	depends on MIPS_ALCHEMY
 	help
 	  This option enables support for the Programmable Serial
 	  Controllers in AC97 and I2S mode, and the Descriptor-Based DMA
-	  Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC.
+	  Controller (DBDMA) as found on the Au12xx/Au13xx/Au1550 SoC.
 
 config SND_SOC_AU1XPSC_I2S
 	tristate
@@ -51,12 +51,14 @@ config SND_SOC_DB1000
 	  of boards (DB1000/DB1500/DB1100).
 
 config SND_SOC_DB1200
-	tristate "DB1200 AC97+I2S audio support"
+	tristate "DB1200/DB1300/DB1550 Audio support"
 	depends on SND_SOC_AU1XPSC
 	select SND_SOC_AU1XPSC_AC97
 	select SND_SOC_AC97_CODEC
+	select SND_SOC_WM9712
 	select SND_SOC_AU1XPSC_I2S
 	select SND_SOC_WM8731
 	help
-	  Select this option to enable audio (AC97 or I2S) on the
-	  Alchemy/AMD/RMI DB1200 demoboard.
+	  Select this option to enable audio (AC97 and I2S) on the
+	  Alchemy/AMD/RMI/NetLogic Db1200, Db1550 and Db1300 evaluation boards.
+	  If you need Db1300 touchscreen support, you definitely want to say Y.
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 1c62939..30ea513 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -1,5 +1,5 @@
 /*
- * DB1200 ASoC audio fabric support code.
+ * DB1200/DB1300/DB1550 ASoC audio fabric support code.
  *
  * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com>
  *
@@ -28,6 +28,18 @@ static struct platform_device_id db1200_pids[] = {
 	}, {
 		.name		= "db1200-i2s",
 		.driver_data	= 1,
+	}, {
+		.name		= "db1300-ac97",
+		.driver_data	= 2,
+	}, {
+		.name		= "db1300-i2s",
+		.driver_data	= 3,
+	}, {
+		.name		= "db1550-ac97",
+		.driver_data	= 4,
+	}, {
+		.name		= "db1550-i2s",
+		.driver_data	= 5,
 	},
 	{},
 };
@@ -50,6 +62,27 @@ static struct snd_soc_card db1200_ac97_machine = {
 	.num_links	= 1,
 };
 
+static struct snd_soc_dai_link db1300_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.codec_dai_name	= "wm9712-hifi",
+	.cpu_dai_name	= "au1xpsc_ac97.1",
+	.platform_name	= "au1xpsc-pcm.1",
+	.codec_name	= "wm9712-codec.1",
+};
+
+static struct snd_soc_card db1300_ac97_machine = {
+	.name		= "DB1300_AC97",
+	.dai_link	= &db1300_ac97_dai,
+	.num_links	= 1,
+};
+
+static struct snd_soc_card db1550_ac97_machine = {
+	.name		= "DB1550_AC97",
+	.dai_link	= &db1200_ac97_dai,
+	.num_links	= 1,
+};
+
 /*-------------------------  I2S PART  ---------------------------*/
 
 static int db1200_i2s_startup(struct snd_pcm_substream *substream)
@@ -100,11 +133,47 @@ static struct snd_soc_card db1200_i2s_machine = {
 	.num_links	= 1,
 };
 
+static struct snd_soc_dai_link db1300_i2s_dai = {
+	.name		= "WM8731",
+	.stream_name	= "WM8731 PCM",
+	.codec_dai_name	= "wm8731-hifi",
+	.cpu_dai_name	= "au1xpsc_i2s.2",
+	.platform_name	= "au1xpsc-pcm.2",
+	.codec_name	= "wm8731.0-001b",
+	.ops		= &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1300_i2s_machine = {
+	.name		= "DB1300_I2S",
+	.dai_link	= &db1300_i2s_dai,
+	.num_links	= 1,
+};
+
+static struct snd_soc_dai_link db1550_i2s_dai = {
+	.name		= "WM8731",
+	.stream_name	= "WM8731 PCM",
+	.codec_dai_name	= "wm8731-hifi",
+	.cpu_dai_name	= "au1xpsc_i2s.3",
+	.platform_name	= "au1xpsc-pcm.3",
+	.codec_name	= "wm8731.0-001b",
+	.ops		= &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1550_i2s_machine = {
+	.name		= "DB1550_I2S",
+	.dai_link	= &db1550_i2s_dai,
+	.num_links	= 1,
+};
+
 /*-------------------------  COMMON PART  ---------------------------*/
 
 static struct snd_soc_card *db1200_cards[] __devinitdata = {
 	&db1200_ac97_machine,
 	&db1200_i2s_machine,
+	&db1300_ac97_machine,
+	&db1300_i2s_machine,
+	&db1550_ac97_machine,
+	&db1550_i2s_machine,
 };
 
 static int __devinit db1200_audio_probe(struct platform_device *pdev)
@@ -138,5 +207,5 @@ static struct platform_driver db1200_audio_driver = {
 module_platform_driver(db1200_audio_driver);
 
 MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("DB1200 ASoC audio support");
+MODULE_DESCRIPTION("DB1200/DB1300/DB1550 ASoC audio support");
 MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d7bd918..f8863eb 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1457,5 +1457,5 @@ static void __exit sgtl5000_exit(void)
 module_exit(sgtl5000_exit);
 
 MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
-MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng@freescale.com>");
+MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 2b40c93..7c7fd92 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -444,6 +444,12 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	/* Enable the FLL */
 	snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA);
 
+	/* Both overestimates */
+	if (Fref < 1000000)
+		msleep(3);
+	else
+		msleep(1);
+
 	dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
 
 	wm8993->fll_fref = Fref;
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 3fc9613..de83904 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -113,9 +113,9 @@ static int ep93xx_pcm_open(struct snd_pcm_substream *substream)
 	rtd->dma_data.name = dma_params->name;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		rtd->dma_data.direction = DMA_TO_DEVICE;
+		rtd->dma_data.direction = DMA_MEM_TO_DEV;
 	else
-		rtd->dma_data.direction = DMA_FROM_DEVICE;
+		rtd->dma_data.direction = DMA_DEV_TO_MEM;
 
 	rtd->dma_chan = dma_request_channel(mask, ep93xx_pcm_dma_filter,
 					    &rtd->dma_data);
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 1cf2fe8..5780c9b 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -88,11 +88,13 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
 	iprtd->dma_data.dma_request = dma_params->dma;
 
 	/* Try to grab a DMA channel */
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
-	if (!iprtd->dma_chan)
-		return -EINVAL;
+	if (!iprtd->dma_chan) {
+		dma_cap_zero(mask);
+		dma_cap_set(DMA_SLAVE, mask);
+		iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+		if (!iprtd->dma_chan)
+			return -EINVAL;
+	}
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
@@ -107,12 +109,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
 	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		slave_config.direction = DMA_TO_DEVICE;
+		slave_config.direction = DMA_MEM_TO_DEV;
 		slave_config.dst_addr = dma_params->dma_addr;
 		slave_config.dst_addr_width = buswidth;
 		slave_config.dst_maxburst = dma_params->burstsize;
 	} else {
-		slave_config.direction = DMA_FROM_DEVICE;
+		slave_config.direction = DMA_DEV_TO_MEM;
 		slave_config.src_addr = dma_params->dma_addr;
 		slave_config.src_addr_width = buswidth;
 		slave_config.src_maxburst = dma_params->burstsize;
@@ -159,7 +161,7 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
 			iprtd->period_bytes * iprtd->periods,
 			iprtd->period_bytes,
 			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-			DMA_TO_DEVICE : DMA_FROM_DEVICE);
+			DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
 	if (!iprtd->desc) {
 		dev_err(&chan->dev->device, "cannot prepare slave dma\n");
 		return -EINVAL;
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 0e12f4e..105f42a 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -136,7 +136,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream,
 			iprtd->period_bytes * iprtd->periods,
 			iprtd->period_bytes,
 			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-			DMA_TO_DEVICE : DMA_FROM_DEVICE);
+			DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
 	if (!iprtd->desc) {
 		dev_err(&chan->dev->device, "cannot prepare slave dma\n");
 		return -EINVAL;
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 45d11dd..946020a 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -366,7 +366,7 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
 		goto out3;
 
 	/* enbale ac97 multifunction pin */
-	mfp_set_groupg(nuc900_audio->dev, "nuc900-audio");
+	mfp_set_groupg(nuc900_audio->dev, NULL);
 
 	return 0;
 
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 427ae0d..e4ba17c 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -86,7 +86,7 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
 	dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE);
 	dma_info.direction =
 		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK
-		? DMA_TO_DEVICE : DMA_FROM_DEVICE);
+		? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
 	dma_info.fp = audio_buffdone;
 	dma_info.fp_param = substream;
 	dma_info.period = prtd->dma_period;
@@ -171,7 +171,7 @@ static int dma_hw_params(struct snd_pcm_substream *substream,
 		dma_info.client = prtd->params->client;
 		dma_info.direction =
 			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK
-			? DMA_TO_DEVICE : DMA_FROM_DEVICE);
+			? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
 		dma_info.width = prtd->params->dma_size;
 		dma_info.fifo = prtd->params->dma_addr;
 		prtd->params->ch = prtd->params->ops->request(
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index f8f6816..0193e59 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -131,7 +131,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info,
 	sg_dma_address(&sg) = buff;
 
 	desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan,
-		&sg, 1, DMA_TO_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+		&sg, 1, DMA_MEM_TO_DEV, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
 	if (!desc) {
 		dev_err(dev, "Failed to allocate a dma descriptor\n");
 		return -ENOMEM;
@@ -181,7 +181,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info,
 	sg_dma_address(&sg) = buff;
 
 	desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan,
-		&sg, 1, DMA_FROM_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+		&sg, 1, DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
 	if (!desc) {
 		dev_err(dev, "Failed to allocate dma descriptor\n");
 		return -ENOMEM;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3986520..b5ecf6d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -907,6 +907,10 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 			if (err < 0)
 				printk(KERN_ERR "asoc: failed to remove %s\n", platform->name);
 		}
+
+		/* Make sure all DAPM widgets are freed */
+		snd_soc_dapm_free(&platform->dapm);
+
 		platform->probed = 0;
 		list_del(&platform->card_list);
 		module_put(platform->dev->driver->owner);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 3ad1f59..1f55ded 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1426,7 +1426,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 			dapm->target_bias_level = SND_SOC_BIAS_ON;
 			break;
 		case SND_SOC_DAPM_STREAM_STOP:
-			if (dapm->codec->active)
+			if (dapm->codec && dapm->codec->active)
 				dapm->target_bias_level = SND_SOC_BIAS_ON;
 			else
 				dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 93931de..2155461 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -134,7 +134,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
 	sg_dma_address(&sg) = buf_dma_addr;
 	desc = chan->device->device_prep_slave_sg(chan, &sg, 1,
 		dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-		DMA_TO_DEVICE : DMA_FROM_DEVICE,
+		DMA_MEM_TO_DEV : DMA_DEV_TO_MEM,
 		DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
 	if (!desc) {
 		dev_err(&chan->dev->device, "cannot prepare slave dma\n");