Merge branch 'fix/pcm-jiffies-check' into topic/pcm-jiffies-check

31 files changed, 225 insertions, 103 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 7fbd68fab94..5c48e36038f 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 a2c12d105c9..6fdca97186e 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 63d088f2265..d659995ac3a 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,17 @@ 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.
+	 */
+	if (runtime->hw.info & SNDRV_PCM_INFO_BATCH)
+		goto no_jiffies_check;
 	hdelta = new_hw_ptr - old_hw_ptr;
 	jdelta = jiffies - runtime->hw_ptr_jiffies;
 	if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
@@ -272,6 +283,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 		hw_base -= hw_base % runtime->buffer_size;
 		delta = 0;
 	}
+ no_jiffies_check:
 	if (delta > runtime->period_size + runtime->period_size / 2) {
 		hw_ptr_error(substream,
 			     "Lost interrupts? "
@@ -329,7 +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",
@@ -1471,7 +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 cb769d415db..45dc53fcfa2 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -849,6 +849,7 @@ 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)
@@ -962,6 +963,11 @@ 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 caeb0f57fcc..199b0337714 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 b2b6d50c942..a25fb7b1f44 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/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
index 906454413ed..3a1526ae172 100644
--- a/sound/isa/msnd/msnd.c
+++ b/sound/isa/msnd/msnd.c
@@ -438,7 +438,8 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip,
 static struct snd_pcm_hardware snd_msnd_playback = {
 	.info =			SNDRV_PCM_INFO_MMAP |
 				SNDRV_PCM_INFO_INTERLEAVED |
-				SNDRV_PCM_INFO_MMAP_VALID,
+				SNDRV_PCM_INFO_MMAP_VALID |
+				SNDRV_PCM_INFO_BATCH,
 	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =		SNDRV_PCM_RATE_8000_48000,
 	.rate_min =		8000,
@@ -456,7 +457,8 @@ static struct snd_pcm_hardware snd_msnd_playback = {
 static struct snd_pcm_hardware snd_msnd_capture = {
 	.info =			SNDRV_PCM_INFO_MMAP |
 				SNDRV_PCM_INFO_INTERLEAVED |
-				SNDRV_PCM_INFO_MMAP_VALID,
+				SNDRV_PCM_INFO_MMAP_VALID |
+				SNDRV_PCM_INFO_BATCH,
 	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =		SNDRV_PCM_RATE_8000_48000,
 	.rate_min =		8000,
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 81bc93e5f1e..7337abdbe4e 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/bt87x.c b/sound/pci/bt87x.c
index a299340519d..ce3f2e90f4d 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -349,7 +349,8 @@ static struct snd_pcm_hardware snd_bt87x_digital_hw = {
 	.info = SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP_VALID,
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	.rates = 0, /* set at runtime */
 	.channels_min = 2,
@@ -365,7 +366,8 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = {
 	.info = SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP_VALID,
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
 	.rates = SNDRV_PCM_RATE_KNOT,
 	.rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX,
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index ad2888705d2..c111efe61c3 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/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
index 3482ef69f49..2e44316530a 100644
--- a/sound/pci/echoaudio/indigodjx.c
+++ b/sound/pci/echoaudio/indigodjx.c
@@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
 	.rates = 	SNDRV_PCM_RATE_32000 |
 			SNDRV_PCM_RATE_44100 |
 			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_64000 |
 			SNDRV_PCM_RATE_88200 |
 			SNDRV_PCM_RATE_96000,
 	.rate_min = 32000,
diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c
index aebee27a40f..eb3819f9654 100644
--- a/sound/pci/echoaudio/indigoiox.c
+++ b/sound/pci/echoaudio/indigoiox.c
@@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
 	.rates = 	SNDRV_PCM_RATE_32000 |
 			SNDRV_PCM_RATE_44100 |
 			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_64000 |
 			SNDRV_PCM_RATE_88200 |
 			SNDRV_PCM_RATE_96000,
 	.rate_min = 32000,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8a0d3e7927..bcbb736f94f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12058,6 +12058,7 @@ 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 917bc5d3ac2..03b3646018a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4079,7 +4079,12 @@ 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/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8b79969034b..7cc38a11e99 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1238,7 +1238,8 @@ static struct snd_pcm_hardware snd_korg1212_playback_info =
 {
 	.info =              (SNDRV_PCM_INFO_MMAP |
                               SNDRV_PCM_INFO_MMAP_VALID |
-                              SNDRV_PCM_INFO_INTERLEAVED),
+			      SNDRV_PCM_INFO_INTERLEAVED |
+			      SNDRV_PCM_INFO_BATCH),
 	.formats =	      SNDRV_PCM_FMTBIT_S16_LE,
         .rates =              (SNDRV_PCM_RATE_44100 |
                               SNDRV_PCM_RATE_48000),
@@ -1258,7 +1259,8 @@ static struct snd_pcm_hardware snd_korg1212_capture_info =
 {
         .info =              (SNDRV_PCM_INFO_MMAP |
                               SNDRV_PCM_INFO_MMAP_VALID |
-                              SNDRV_PCM_INFO_INTERLEAVED),
+			      SNDRV_PCM_INFO_INTERLEAVED |
+			      SNDRV_PCM_INFO_BATCH),
         .formats =	      SNDRV_PCM_FMTBIT_S16_LE,
         .rates =	      (SNDRV_PCM_RATE_44100 |
                               SNDRV_PCM_RATE_48000),
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 6f1034417a0..e51a5ef1954 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 809b233dd4a..1ef58c51c21 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/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 01066c95580..d057e648964 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -240,7 +240,8 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
 static struct snd_pcm_hardware pdacf_pcm_capture_hw = {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
-				 SNDRV_PCM_INFO_MMAP_VALID),
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_BATCH),
 	.formats =		SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
 				SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
 				SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a25914..594c6c5b783 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
 	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-			    SNDRV_PCM_INFO_INTERLEAVED,
+			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
 	.formats	  = AU1XPSC_PCM_FMTS,
 	.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
 	.period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28..df7c8c281d2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		4, 3, 0, output_tvl),
 
 	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+		TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
 	/* Common capture gain controls */
 	SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed..0275321ff8a 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
 	 * required for LRC in master mode. The DACs or ADCs need a
 	 * valid audio path i.e. pin -> ADC or DAC -> pin before
 	 * the LRC will be enabled in master mode. */
-	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+	if (!master || cmd != SNDRV_PCM_TRIGGER_START)
 		return 0;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3..40cd274eb1e 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 bd7392c9657..411a710be66 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 9b90b347007..58fd1cbedd8 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 ffdb9439d3d..b1ea52fc83c 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 *p