33 files changed, 267 insertions, 112 deletions

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 63d088f2265f..a2a792c18c40 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,12 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
                         new_hw_ptr = hw_base + pos;
                 }
         }
+        /* 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 +278,7 @@ 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? "
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index caeb0f57fcca..771955a9be71 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,7 +50,7 @@ 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",
+        sprintf(uinfo->value.enumerated.name, "%lu",
                         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/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
index 906454413ed2..3a1526ae1729 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/bt87x.c b/sound/pci/bt87x.c
index a299340519df..ce3f2e90f4d7 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/cmipci.c b/sound/pci/cmipci.c
index c7899c32aba1..449fe02f666e 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3014,7 +3014,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
                 .dev_free =     snd_cmipci_dev_free,
         };
         unsigned int val;
-        long iomidi;
+        long iomidi = 0;
         int integrated_midi = 0;
         char modelstr[16];
         int pcm_index, pcm_spdif_index;
diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
index 3482ef69f491..2e44316530a2 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 aebee27a40ff..eb3819f9654a 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_analog.c b/sound/pci/hda/patch_analog.c
index 9bcd8ab5a27f..84cc49ca9148 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3817,6 +3817,49 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
         { } /* end */
 };
 
+static struct hda_verb ad1884a_mobile_verbs[] = {
+        /* DACs; unmute as default */
+        {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+        {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+        /* Port-A (HP) mixer - route only from analog mixer */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        /* Port-A pin */
+        {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+        /* Port-A (HP) pin - always unmuted */
+        {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        /* Port-B (mic jack) pin */
+        {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+        /* Port-C (int mic) pin */
+        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+        {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+        /* Port-F (int speaker) mixer - route only from analog mixer */
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        /* Port-F pin */
+        {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+        {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* Analog mixer; mute as default */
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+        /* Analog Mix output amp */
+        {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* capture sources */
+        /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* unsolicited event for pin-sense */
+        {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+        { } /* end */
+};
+
 /*
  * Thinkpad X300
  * 0x11 - HP
@@ -3988,7 +4031,7 @@ static int patch_ad1884a(struct hda_codec *codec)
                 break;
         case AD1884A_MOBILE:
                 spec->mixers[0] = ad1884a_mobile_mixers;
-                spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
+                spec->init_verbs[0] = ad1884a_mobile_verbs;
                 spec->multiout.dig_out_nid = 0;
                 codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
                 codec->patch_ops.init = ad1884a_hp_init;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 917bc5d3ac2c..03b3646018a1 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 8b79969034be..7cc38a11e997 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 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/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 01066c95580e..d057e6489643 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 30490a259148..594c6c5b7838 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/Makefile b/sound/soc/codecs/Makefile
index 030d2454725f..f2653803ede8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)    += snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)    += snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8990)    += snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991)    += snd-soc-wm8991.o
 obj-$(CONFIG_SND_SOC_WM9705)    += snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)    += snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)    += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..df7c8c281d2f 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 3b1d0993bed9..0275321ff8ab 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/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f160fc..9f6be3d31ac0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
                          struct snd_ctl_elem_value *ucontrol)
 {
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-        int reg = kcontrol->private_value & 0xff;
-        int reg2 = (kcontrol->private_value >> 24) & 0xff;
+        unsigned int reg = mc->reg;
+        unsigned int reg2 = mc->rreg;
         int ret;
         u16 val;
 
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
         return 0;
 }
 
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+                                    xinvert, tlv_array)                 \
 {       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
         .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
                 SNDRV_CTL_ELEM_ACCESS_READWRITE,  \
         .tlv.p = (tlv_array), \
         .info = snd_soc_info_volsw_2r, \
         .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
-        .private_value = (reg_left) | ((shift) << 8)  |         \
-                ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+        .private_value = (unsigned long)&(struct soc_mixer_control) \
+                {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+                .max = xmax, .invert = xinvert} }
 
 static const struct snd_kcontrol_new wm8580_snd_controls[] = {
 SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
         reg = wm8580_read(codec, WM8580_PLLA4 + offset);
         reg &= ~0x3f;
         reg |= pll_div.prescale | pll_div.postscale << 1 |
-                pll_div.freqmode << 4;
+                pll_div.freqmode << 3;
 
         wm8580_write(codec, WM8580_PLLA4 + offset, reg);
 
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/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..1111c710118a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 
 static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
         .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-                SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+                SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_BATCH,
         .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
                    SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
         .rate_min = 8000,
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 689ffcd17e1f..ab680aac3fcb 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -636,5 +636,6 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 
         return snd_soc_register_dai(dai);
 }
-
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
         .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 =      DMABRG_FMTS,
         .rates =        DMABRG_RATES,
         .rate_min =             8000,
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/sparc/dbri.c b/sound/sparc/dbri.c
index af95ff1e126c..1d2e51b3f918 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -1975,7 +1975,8 @@ static struct snd_pcm_hardware snd_dbri_pcm_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_MU_LAW |
                           SNDRV_PCM_FMTBIT_A_LAW |
                           SNDRV_PCM_FMTBIT_U8 |
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 3f45c0fe61ab..b13ce767ac72 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -195,11 +195,14 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
 
         debug("%s(%p)\n", __func__, substream);
         
-        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                dev->period_out_count[index] = BYTES_PER_SAMPLE + 1;
                 dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
-        else
+        } else {
+                dev->period_in_count[index] = BYTES_PER_SAMPLE;
                 dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE;
-        
+        }
+
         if (dev->streaming)
                 return 0;
         
@@ -300,8 +303,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev,
                 if (!sub)
                         continue;
 
-                pb = frames_to_bytes(sub->runtime, 
-                                     sub->runtime->period_size);
+                pb = snd_pcm_lib_period_bytes(sub);
                 cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
                                         &dev->period_out_count[stream] :
                                         &dev->period_in_count[stream];
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 6d517705da0e..515de1cd2a3e 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
                          "{Native Instruments, RigKontrol3},"
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 9a608fa85155..dd1ab6177840 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -870,7 +870,8 @@ static struct snd_pcm_hardware snd_usX2Y_2c =
 {
         .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_S24_3LE,
         .rates =                   SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
         .rate_min =                44100,