Merge branch 'next-mtd' of git://aeryn.fluff.org.uk/bjdooks/linux

24 files changed, 217 insertions, 93 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 7fbd68fab944..5c48e36038f2 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -1074,7 +1074,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
 	return i;
 }
 
-static int __devinit aaci_probe(struct amba_device *dev, void *id)
+static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
 {
 	struct aaci *aaci;
 	int ret, i;
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index a2c12d105c9a..6fdca97186e7 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action)
 		switch (resetgpio_action) {
 		case RESETGPIO_NORMAL_ALTFUNC:
 			if (reset_gpio == 113)
-				mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
+				mode = 113 | GPIO_ALT_FN_2_OUT;
 			if (reset_gpio == 95)
 				mode = 95 | GPIO_ALT_FN_1_OUT;
 			break;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a2a792c18c40..d659995ac3ac 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,11 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 			new_hw_ptr = hw_base + pos;
 		}
 	}
+
+	/* Do jiffies check only in xrun_debug mode */
+	if (!xrun_debug(substream))
+		goto no_jiffies_check;
+
 	/* Skip the jiffies check for hardwares with BATCH flag.
 	 * Such hardware usually just increases the position at each IRQ,
 	 * thus it can't give any strange position.
@@ -336,7 +341,9 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
 			hw_base = 0;
 		new_hw_ptr = hw_base + pos;
 	}
-	if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
+	/* Do jiffies check only in xrun_debug mode */
+	if (xrun_debug(substream) &&
+	    ((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
 		hw_ptr_error(substream,
 			     "hw_ptr skipping! "
 			     "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n",
@@ -1478,7 +1485,6 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream,
 		runtime->status->hw_ptr %= runtime->buffer_size;
 	else
 		runtime->status->hw_ptr = 0;
-	runtime->hw_ptr_jiffies = jiffies;
 	snd_pcm_stream_unlock_irqrestore(substream, flags);
 	return 0;
 }
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index fc6f98e257df..b5da656d1ece 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -848,6 +848,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_trigger_tstamp(substream);
+	runtime->hw_ptr_jiffies = jiffies;
 	runtime->status->state = state;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
 	    runtime->silence_size > 0)
@@ -961,6 +962,11 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
 {
 	if (substream->runtime->trigger_master != substream)
 		return 0;
+	/* The jiffies check in snd_pcm_update_hw_ptr*() is done by
+	 * a delta betwen the current jiffies, this gives a large enough
+	 * delta, effectively to skip the check once.
+	 */
+	substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000;
 	return substream->ops->trigger(substream,
 				       push ? SNDRV_PCM_TRIGGER_PAUSE_PUSH :
 					      SNDRV_PCM_TRIGGER_PAUSE_RELEASE);
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index caeb0f57fcca..199b03377142 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,8 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
 	uinfo->value.enumerated.items = chip->max_treble + 1;
 	if (uinfo->value.enumerated.item > chip->max_treble)
 		uinfo->value.enumerated.item = chip->max_treble;
-	sprintf(uinfo->value.enumerated.name, "%d",
-			PCSP_CALC_RATE(uinfo->value.enumerated.item));
+	sprintf(uinfo->value.enumerated.name, "%lu",
+		(unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item));
 	return 0;
 }
 
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index b2b6d50c9425..a25fb7b1f441 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -963,16 +963,11 @@ static int __devinit snd_serial_probe(struct platform_device *devptr)
 	if (err < 0)
 		goto _err;
 
-	sprintf(card->longname, "%s at 0x%lx, irq %d speed %d div %d outs %d ins %d adaptor %s droponfull %d",
+	sprintf(card->longname, "%s [%s] at %#lx, irq %d",
 		card->shortname,
-		uart->base,
-		uart->irq,
-		uart->speed,
-		(int)uart->divisor,
-		outs[dev],
-		ins[dev],
 		adaptor_names[uart->adaptor],
-		uart->drop_on_full);
+		uart->base,
+		uart->irq);
 
 	snd_card_set_dev(card, &devptr->dev);
 
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 81bc93e5f1e3..7337abdbe4e3 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97)
 }
 
 static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker =
-AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch",
+		AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
 
+/* "Sigmatel " removed due to excessive name length: */
 static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert =
-AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
+AC97_SINGLE("Surround Phase Inversion Playback Switch",
+		AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
 
 static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = {
 AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0),
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index ad2888705d2a..c111efe61c3c 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
 		"Capture Volume",
 		"External Amplifier",
 		"Sigmatel 4-Speaker Stereo Playback Switch",
-		"Sigmatel Surround Phase Inversion Playback ",
+		"Surround Phase Inversion Playback Switch",
 		NULL
 	};
 	static char *ca0106_rename_ctls[] = {
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 21e99cfa8c49..3128e1a6bc65 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2141,6 +2141,7 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
 	SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
 	/* forced codec slots */
+	SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103),
 	SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
 	{}
 };
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 56ce19e68cb5..4fcbe21829ab 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1848,6 +1848,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = {
 
 static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736),
+	SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP),
 	SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
 		      CXT5051_LAPTOP),
 	SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8a0d3e79272..0fd258eba3a5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -776,6 +776,12 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
 		pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
 		if (pincap & AC_PINCAP_VREF_80)
 			val = PIN_VREF80;
+		else if (pincap & AC_PINCAP_VREF_50)
+			val = PIN_VREF50;
+		else if (pincap & AC_PINCAP_VREF_100)
+			val = PIN_VREF100;
+		else if (pincap & AC_PINCAP_VREF_GRD)
+			val = PIN_VREFGRD;
 	}
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
 }
@@ -12058,6 +12064,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
 	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
 	SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+	SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 917bc5d3ac2c..d2fd8ef6aef8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -150,6 +150,7 @@ enum {
 	STAC_D965_REF,
 	STAC_D965_3ST,
 	STAC_D965_5ST,
+	STAC_D965_5ST_NO_FP,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
 	STAC_927X_MODELS
@@ -2154,6 +2155,13 @@ static unsigned int d965_5st_pin_configs[14] = {
 	0x40000100, 0x40000100
 };
 
+static unsigned int d965_5st_no_fp_pin_configs[14] = {
+	0x40000100, 0x40000100, 0x0181304e, 0x01014010,
+	0x01a19040, 0x01011012, 0x01016011, 0x40000100,
+	0x40000100, 0x40000100, 0x40000100, 0x01442070,
+	0x40000100, 0x40000100
+};
+
 static unsigned int dell_3st_pin_configs[14] = {
 	0x02211230, 0x02a11220, 0x01a19040, 0x01114210,
 	0x01111212, 0x01116211, 0x01813050, 0x01112214,
@@ -2166,6 +2174,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 	[STAC_D965_REF]  = ref927x_pin_configs,
 	[STAC_D965_3ST]  = d965_3st_pin_configs,
 	[STAC_D965_5ST]  = d965_5st_pin_configs,
+	[STAC_D965_5ST_NO_FP]  = d965_5st_no_fp_pin_configs,
 	[STAC_DELL_3ST]  = dell_3st_pin_configs,
 	[STAC_DELL_BIOS] = NULL,
 };
@@ -2176,6 +2185,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
 	[STAC_D965_REF]		= "ref",
 	[STAC_D965_3ST]		= "3stack",
 	[STAC_D965_5ST]		= "5stack",
+	[STAC_D965_5ST_NO_FP]	= "5stack-no-fp",
 	[STAC_DELL_3ST]		= "dell-3stack",
 	[STAC_DELL_BIOS]	= "dell-bios",
 };
@@ -4079,7 +4089,12 @@ static int stac92xx_init(struct hda_codec *codec)
 				pinctl = snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 				/* if PINCTL already set then skip */
-				if (!(pinctl & AC_PINCTL_IN_EN)) {
+				/* Also, if both INPUT and OUTPUT are set,
+				 * it must be a BIOS bug; need to override, too
+				 */
+				if (!(pinctl & AC_PINCTL_IN_EN) ||
+				    (pinctl & AC_PINCTL_OUT_EN)) {
+					pinctl &= ~AC_PINCTL_OUT_EN;
 					pinctl |= AC_PINCTL_IN_EN;
 					stac92xx_auto_set_pinctl(codec, nid,
 								 pinctl);
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 6f1034417a02..e51a5ef1954d 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -889,7 +889,7 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
 	spin_lock_irqsave(&cif->lock, irqflags);
 	while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport))
 		udelay(10);
-	if (i >= CMDIF_TIMEOUT) {
+	if (i > CMDIF_TIMEOUT) {
 		err = -EBUSY;
 		goto errout;
 	}
@@ -907,8 +907,10 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
 			WRITE_PORT_ULONG(cmdport->data1, cmd);	/* write cmd */
 			if ((flags & RESP) && ret) {
 				while (!IS_DATF(cmdport) &&
-				       time++ < CMDIF_TIMEOUT)
+				       time < CMDIF_TIMEOUT) {
 					udelay(10);
+					time++;
+				}
 				if (time < CMDIF_TIMEOUT) {	/* read response */
 					ret->retlongs[0] =
 					    READ_PORT_ULONG(cmdport->data1);
@@ -1454,7 +1456,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd)
 			SEND_GPOS(cif, 0, data->id, &rptr);
 			udelay(1);
 		} while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY);
-		if (j >= MAX_WRITE_RETRY)
+		if (j > MAX_WRITE_RETRY)
 			snd_printk(KERN_ERR "Riptide: Could not stop stream!");
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -1783,7 +1785,7 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg,
 		SEND_SACR(cif, val, reg);
 		SEND_RACR(cif, reg, &rptr);
 	} while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY);
-	if (i == MAX_WRITE_RETRY)
+	if (i > MAX_WRITE_RETRY)
 		snd_printdd("Write AC97 reg failed\n");
 }
 
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 809b233dd4a3..1ef58c51c213 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1687,7 +1687,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1);
 
 static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = {
 	.name = "PCM Playback Volume",
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3f..40cd274eb1ef 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
 	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
 	&wm8990_dapm_ainlmux_controls, inmixer_event,
 	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
 	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
 	&wm8990_dapm_ainrmux_controls, inmixer_event,
 	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
 	/* LIN34 PGA */
 	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
-	{"LIN34 PGA", "LIN4 Switch", "LIN4"},
+	{"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
 	/* INMIXL */
 	{"INMIXL", "Record Left Volume", "LOMIX"},
 	{"INMIXL", "LIN2 Volume", "LIN2"},
 	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
 	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
-	/* AILNMUX */
-	{"AILNMUX", "INMIXL Mix", "INMIXL"},
-	{"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
-	{"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
-	{"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
-	{"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* AINLMUX */
+	{"AINLMUX", "INMIXL Mix", "INMIXL"},
+	{"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+	{"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+	{"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
 	/* ADC */
-	{"Left ADC", NULL, "AILNMUX"},
+	{"Left ADC", NULL, "AINLMUX"},
 
 	/* RIN12 PGA */
 	{"RIN12 PGA", "RIN1 Switch", "RIN1"},
 	{"RIN12 PGA", "RIN2 Switch", "RIN2"},
 	/* RIN34 PGA */
 	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
-	{"RIN34 PGA", "RIN4 Switch", "RIN4"},
+	{"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
 	/* INMIXL */
 	{"INMIXR", "Record Right Volume", "ROMIX"},
 	{"INMIXR", "RIN2 Volume", "RIN2"},
 	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
 	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
-	/* AIRNMUX */
-	{"AIRNMUX", "INMIXR Mix", "INMIXR"},
-	{"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
-	{"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
-	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
-	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* AINRMUX */
+	{"AINRMUX", "INMIXR Mix", "INMIXR"},
+	{"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+	{"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+	{"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
 	/* ADC */
-	{"Right ADC", NULL, "AIRNMUX"},
+	{"Right ADC", NULL, "AINRMUX"},
 
 	/* LOMIX */
 	{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
 
 	/* OUT3MIX */
-	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
 	{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
 
 	/* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	/* Output Pins */
 	{"LON", NULL, "LONMIX"},
 	{"LOP", NULL, "LOPMIX"},
-	{"OUT", NULL, "OUT3MIX"},
+	{"OUT3", NULL, "OUT3MIX"},
 	{"LOUT", NULL, "LOUT PGA"},
 	{"SPKN", NULL, "SPKMIX"},
 	{"ROUT", NULL, "ROUT PGA"},
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c9657e..411a710be660 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
 	tristate
 
 config SND_DAVINCI_SOC_EVM
-	tristate "SoC Audio support for DaVinci EVM"
-	depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+	tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+	depends on SND_DAVINCI_SOC
+	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
 	select SND_DAVINCI_SOC_I2S
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on TI
-	  DaVinci EVM platform.
+	  DaVinci DM6446 or DM355 EVM platforms.
 
 config SND_DAVINCI_SOC_SFFSDR
 	tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b347007c..58fd1cbedd88 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
 
 /* evm audio private data */
 static struct aic3x_setup_data evm_aic3x_setup = {
-	.i2c_bus = 0,
+	.i2c_bus = 1,
 	.i2c_address = 0x1b,
 };
 
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
 	.codec_data = &evm_aic3x_setup,
 };
 
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
 static struct resource evm_snd_resources[] = {
 	{
-		.start = DAVINCI_MCBSP_BASE,
-		.end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+		.start = DAVINCI_ASP0_BASE,
+		.end = DAVINCI_ASP0_BASE + SZ_8K - 1,
 		.flags = IORESOURCE_MEM,
 	},
 };
 
 static struct evm_snd_platform_data evm_snd_data = {
-	.tx_dma_ch	= DM644X_DMACH_MCBSP_TX,
-	.rx_dma_ch	= DM644X_DMACH_MCBSP_RX,
+	.tx_dma_ch	= DAVINCI_DMA_ASP0_TX,
+	.rx_dma_ch	= DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+	{
+		.start = DAVINCI_ASP1_BASE,
+		.end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+		.flags = IORESOURCE_MEM,
+	},
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+	.tx_dma_ch	= DAVINCI_DMA_ASP1_TX,
+	.rx_dma_ch	= DAVINCI_DMA_ASP1_RX,
 };
 
 static struct platform_device *evm_snd_device;
 
 static int __init evm_init(void)
 {
+	struct resource *resources;
+	unsigned num_resources;
+	struct evm_snd_platform_data *data;
+	int index;
 	int ret;
 
-	evm_snd_device = platform_device_alloc("soc-audio", 0);
+	if (machine_is_davinci_evm()) {
+		davinci_cfg_reg(DM644X_MCBSP);
+
+		resources = evm_snd_resources;
+		num_resources = ARRAY_SIZE(evm_snd_resources);
+		data = &evm_snd_data;
+		index = 0;
+	} else if (machine_is_davinci_dm355_evm()) {
+		/* we don't use ASP1 IRQs, or we'd need to mux them ... */
+		davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+		davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+		resources = dm335evm_snd_resources;
+		num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+		data = &dm335evm_snd_data;
+		index = 1;
+	} else
+		return -EINVAL;
+
+	evm_snd_device = platform_device_alloc("soc-audio", index);
 	if (!evm_snd_device)
 		return -ENOMEM;
 
 	platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
 	evm_snd_devdata.dev = &evm_snd_device->dev;
-	platform_device_add_data(evm_snd_device, &evm_snd_data,
-				 sizeof(evm_snd_data));
+	platform_device_add_data(evm_snd_device, data, sizeof(*data));
 
-	ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
-					    ARRAY_SIZE(evm_snd_resources));
+	ret = platform_device_add_resources(evm_snd_device, resources,
+			num_resources);
 	if (ret) {
 		platform_device_put(evm_snd_device);
 		return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb9439d3d8..b1ea52fc83c7 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
 
 #include "davinci-pcm.h"
 
+
+/*
+ * NOTE:  terminology here is confusing.
+ *
+ *  - This driver supports the "Audio Serial Port" (ASP),
+ *    found on dm6446, dm355, and other DaVinci chips.
+ *
+ *  - But it labels it a "Multi-channel Buffered Serial Port"
+ *    (McBSP) as on older chips like the dm642 ... which was
+ *    backward-compatible, possibly explaining that confusion.
+ *
+ *  - OMAP chips have a controller called McBSP, which is
+ *    incompatible with the DaVinci flavor of McBSP.
+ *
+ *  - Newer DaVinci chips have a controller called McASP,
+ *    incompatible with ASP and with either McBSP.
+ *
+ * In short:  this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
 #define DAVINCI_MCBSP_DRR_REG	0x00
 #define DAVINCI_MCBSP_DXR_REG	0x04
 #define DAVINCI_MCBSP_SPCR_REG	0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *ioarea;
 	struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 
 	cpu_dai->private_data = dev;
 
-	dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+	dev->clk = clk_get(&pdev->dev, NULL);
 	if (IS_ERR(dev->clk)) {
 		ret = -ENODEV;
 		goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
 	struct resource *mem;
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b3a53d..a05996588489 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
+#include <mach/edma.h>
 
 #include "davinci-pcm.h"
 
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
 	spinlock_t lock;
 	int period;		/* current DMA period */
 	int master_lch;		/* Master DMA channel */
-	int slave_lch;		/* Slave DMA channel */
+	int slave_lch;		/* linked parameter RAM reload slot */
 	struct davinci_pcm_dma_params *params;	/* DMA params */
 };
 
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 		dst_bidx = data_type;
 	}
 
-	davinci_set_dma_src_params(lch, src, INCR, W8BIT);
-	davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
-	davinci_set_dma_src_index(lch, src_bidx, 0);
-	davinci_set_dma_dest_index(lch, dst_bidx, 0);
-	davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+	edma_set_src(lch, src, INCR, W8BIT);
+	edma_set_dest(lch, dst, INCR, W8BIT);
+	edma_set_src_index(lch, src_bidx, 0);
+	edma_set_dest_index(lch, dst_bidx, 0);
+	edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
 
 	prtd->period++;
 	if (unlikely(prtd->period >= runtime->periods))
 		prtd->period = 0;
 }
 
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 {
 	struct snd_pcm_substream *substream = data;
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
-	int tcc = TCC_ANY;
+	struct edmacc_param p_ram;
 	int ret;
 
 	if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	prtd->params = dma_data;
 
 	/* Request master DMA channel */
-	ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+	ret = edma_alloc_channel(prtd->params->channel,
 				  davinci_pcm_dma_irq, substream,
-				  &prtd->master_lch, &tcc, EVENTQ_0);
-	if (ret)
+				  EVENTQ_0);
+	if (ret < 0)
 		return ret;
+	prtd->master_lch = ret;
 
-	/* Request slave DMA channel */
-	ret = davinci_request_dma(PARAM_ANY, "Link",
-				  NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
-	if (ret) {
-		davinci_free_dma(prtd->master_lch);
+	/* Request parameter RAM reload slot */
+	ret = edma_alloc_slot(EDMA_SLOT_ANY);
+	if (ret < 0) {
+		edma_free_channel(prtd->master_lch);
 		return ret;
 	}
-
-	/* Link slave DMA channel in loopback */
-	davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+	prtd->slave_lch = ret;
+
+	/* Issue transfer completion IRQ when the channel completes a
+	 * transfer, then always reload from the same slot (by a kind
+	 * of loopback link).  The completion IRQ handler will update
+	 * the reload slot with a new buffer.
+	 *
+	 * REVISIT save p_ram here after setting up everything except
+	 * the buffer and its length (ccnt) ... use it as a template
+	 * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+	 */
+	edma_read_slot(prtd->slave_lch, &p_ram);
+	p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+	p_ram.link_bcntrld = prtd->slave_lch << 5;
+	edma_write_slot(prtd->slave_lch, &p_ram);
 
 	return 0;
 }
@@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		davinci_start_dma(prtd->master_lch);
+		edma_start(prtd->master_lch);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		davinci_stop_dma(prtd->master_lch);
+		edma_stop(prtd->master_lch);
 		break;
 	default:
 		ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct paramentry_descriptor temp;
+	struct edmacc_param temp;
 
 	prtd->period = 0;
 	davinci_pcm_enqueue_dma(substream);
 
-	/* Get slave channel dma params for master channel startup */
-	davinci_get_dma_params(prtd->slave_lch, &temp);
-	davinci_set_dma_params(prtd->master_lch, &temp);
+	/* Copy self-linked parameter RAM entry into master channel */
+	edma_read_slot(prtd->slave_lch, &temp);
+	edma_write_slot(prtd->master_lch, &temp);
 
 	return 0;
 }
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
 
 	spin_lock(&prtd->lock);
 
-	davinci_dma_getposition(prtd->master_lch, &src, &dst);
+	edma_get_position(prtd->master_lch, &src, &dst);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		count = src - runtime->dma_addr;
 	else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct davinci_runtime_data *prtd = runtime->private_data;
 
-	davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+	edma_unlink(prtd->slave_lch);
 
-	davinci_free_dma(prtd->slave_lch);
-	davinci_free_dma(prtd->master_lch);
+	edma_free_slot(prtd->slave_lch);
+	edma_free_channel(prtd->master_lch);
 
 	kfree(prtd);
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f652d0d..1cd149b9ce69 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
+	if (!card->instantiated)
+		return 0;
+
 	run_delayed_work(&card->delayed_work);
 
 	if (platform->remove)
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 823296d7d578..a6b88482637b 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -3347,7 +3347,7 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface,
 		[QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface,
 		[QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface,
-		[QUIRK_MIDI_RAW] = snd_usb_create_midi_interface,
+		[QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface,
 		[QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface,
 		[QUIRK_MIDI_CME] = snd_usb_create_midi_interface,
 		[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 36e4f7a29adc..8e7f78941ba6 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -153,7 +153,7 @@ enum quirk_type {
 	QUIRK_MIDI_YAMAHA,
 	QUIRK_MIDI_MIDIMAN,
 	QUIRK_MIDI_NOVATION,
-	QUIRK_MIDI_RAW,
+	QUIRK_MIDI_FASTLANE,
 	QUIRK_MIDI_EMAGIC,
 	QUIRK_MIDI_CME,
 	QUIRK_MIDI_US122L,
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 26bad373fe65..2fb35cc22a30 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1778,8 +1778,18 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip,
 		umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
 		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
 		break;
-	case QUIRK_MIDI_RAW:
+	case QUIRK_MIDI_FASTLANE:
 		umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
+		/*
+		 * Interface 1 contains isochronous endpoints, but with the same
+		 * numbers as in interface 0.  Since it is interface 1 that the
+		 * USB core has most recently seen, these descriptors are now
+		 * associated with the endpoint numbers.  This will foul up our
+		 * attempts to submit bulk/interrupt URBs to the endpoints in
+		 * interface 0, so we have to make sure that the USB core looks
+		 * again at interface 0 by calling usb_set_interface() on it.
+		 */
+		usb_set_interface(umidi->chip->dev, 0, 0);
 		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
 		break;
 	case QUIRK_MIDI_EMAGIC:
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 647ef5029651..5d955aaad85f 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -1868,7 +1868,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.data = & (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
-				.type = QUIRK_MIDI_RAW
+				.type = QUIRK_MIDI_FASTLANE
 			},
 			{
 				.ifnum = 1,