Merge branch 'linus' into x86/xen

41 files changed, 411 insertions, 241 deletions

diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 5b3274b465eb..199cca3366df 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -26,10 +26,10 @@
 
 #include <asm/irq.h>
 #include <linux/mutex.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 0ede9e4656a8..381094aab235 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -21,8 +21,8 @@
 #include <sound/pcm_params.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
index faeddf3ecedb..b9c51bf8cd71 100644
--- a/sound/arm/sa11xx-uda1341.c
+++ b/sound/arm/sa11xx-uda1341.c
@@ -71,8 +71,8 @@
 #include <linux/pm.h>
 #endif
 
-#include <asm/hardware.h>
-#include <asm/arch/h3600.h>
+#include <mach/hardware.h>
+#include <mach/h3600.h>
 #include <asm/mach-types.h>
 #include <asm/dma.h>
 
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index 558dadbf45f1..e024e4588b82 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -604,6 +604,9 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in
 {
         struct seq_oss_synth *rec;
 
+        if (dev < 0 || dev >= dp->max_synthdev)
+                return -ENXIO;
+
         if (dp->synths[dev].is_midi) {
                 struct midi_info minf;
                 snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf);
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
index bb4a0969f461..725fef0f59a3 100644
--- a/sound/oss/vidc.c
+++ b/sound/oss/vidc.c
@@ -22,7 +22,7 @@
 #include <linux/kernel.h>
 #include <linux/interrupt.h>
 
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/dma.h>
 #include <asm/io.h>
 #include <asm/hardware/iomd.h>
diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S
index 01ccc074cc11..bed34921d176 100644
--- a/sound/oss/vidc_fill.S
+++ b/sound/oss/vidc_fill.S
@@ -11,7 +11,7 @@
  */
 #include <linux/linkage.h>
 #include <asm/assembler.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/hardware/iomd.h>
 
                 .text
diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c
index 88490418f932..c47842fad657 100644
--- a/sound/oss/waveartist.c
+++ b/sound/oss/waveartist.c
@@ -47,7 +47,7 @@
 #include "waveartist.h"
 
 #ifdef CONFIG_ARM
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/mach-types.h>
 #endif
 
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index f7d95b224a98..31f52d3fc21f 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -845,7 +845,7 @@ config SND_VIRTUOSO
         select SND_OXYGEN_LIB
         help
           Say Y here to include support for sound cards based on the
-          Asus AV100/AV200 chips, i.e., Xonar D2, DX and D2X.
+          Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
 
           To compile this driver as a module, choose M here: the module
           will be called snd-virtuoso.
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ef9f072b47fc..a73d6ca0a906 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -101,6 +101,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
                          "{Intel, ICH8},"
                          "{Intel, ICH9},"
                          "{Intel, ICH10},"
+                         "{Intel, PCH},"
                          "{Intel, SCH},"
                          "{ATI, SB450},"
                          "{ATI, SB600},"
@@ -2263,6 +2264,8 @@ static struct pci_device_id azx_ids[] = {
         { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH },
         { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH },
         { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH },
+        /* PCH */
+        { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH },
         /* SCH */
         { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
         /* ATI SB 450/600 */
@@ -2272,6 +2275,7 @@ static struct pci_device_id azx_ids[] = {
         { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI },
         { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI },
         { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI },
+        { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI },
         { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI },
         { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI },
         { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI },
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index add4e87e0b20..b80e725432f0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6437,6 +6437,39 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
         }
 }
 
+static void alc882_auto_init_input_src(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+        const struct hda_input_mux *imux = spec->input_mux;
+        int c;
+
+        for (c = 0; c < spec->num_adc_nids; c++) {
+                hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
+                hda_nid_t nid = spec->capsrc_nids[c];
+                int conns, mute, idx, item;
+
+                conns = snd_hda_get_connections(codec, nid, conn_list,
+                                                ARRAY_SIZE(conn_list));
+                if (conns < 0)
+                        continue;
+                for (idx = 0; idx < conns; idx++) {
+                        /* if the current connection is the selected one,
+                         * unmute it as default - otherwise mute it
+                         */
+                        mute = AMP_IN_MUTE(idx);
+                        for (item = 0; item < imux->num_items; item++) {
+                                if (imux->items[item].index == idx) {
+                                        if (spec->cur_mux[c] == item)
+                                                mute = AMP_IN_UNMUTE(idx);
+                                        break;
+                                }
+                        }
+                        snd_hda_codec_write(codec, nid, 0,
+                                            AC_VERB_SET_AMP_GAIN_MUTE, mute);
+                }
+        }
+}
+
 /* add mic boosts if needed */
 static int alc_auto_add_mic_boost(struct hda_codec *codec)
 {
@@ -6491,6 +6524,7 @@ static void alc882_auto_init(struct hda_codec *codec)
         alc882_auto_init_multi_out(codec);
         alc882_auto_init_hp_out(codec);
         alc882_auto_init_analog_input(codec);
+        alc882_auto_init_input_src(codec);
         if (spec->unsol_event)
                 alc_sku_automute(codec);
 }
@@ -8285,6 +8319,8 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec)
         }
 }
 
+#define alc883_auto_init_input_src      alc882_auto_init_input_src
+
 /* almost identical with ALC880 parser... */
 static int alc883_parse_auto_config(struct hda_codec *codec)
 {
@@ -8315,6 +8351,7 @@ static void alc883_auto_init(struct hda_codec *codec)
         alc883_auto_init_multi_out(codec);
         alc883_auto_init_hp_out(codec);
         alc883_auto_init_analog_input(codec);
+        alc883_auto_init_input_src(codec);
         if (spec->unsol_event)
                 alc_sku_automute(codec);
 }
@@ -9663,6 +9700,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 #define alc262_auto_init_multi_out      alc882_auto_init_multi_out
 #define alc262_auto_init_hp_out         alc882_auto_init_hp_out
 #define alc262_auto_init_analog_input   alc882_auto_init_analog_input
+#define alc262_auto_init_input_src      alc882_auto_init_input_src
 
 
 /* init callback for auto-configuration model -- overriding the default init */
@@ -9672,6 +9710,7 @@ static void alc262_auto_init(struct hda_codec *codec)
         alc262_auto_init_multi_out(codec);
         alc262_auto_init_hp_out(codec);
         alc262_auto_init_analog_input(codec);
+        alc262_auto_init_input_src(codec);
         if (spec->unsol_event)
                 alc_sku_automute(codec);
 }
@@ -13330,6 +13369,8 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
         }
 }
 
+#define alc861vd_auto_init_input_src    alc882_auto_init_input_src
+
 #define alc861vd_idx_to_mixer_vol(nid)          ((nid) + 0x02)
 #define alc861vd_idx_to_mixer_switch(nid)       ((nid) + 0x0c)
 
@@ -13512,6 +13553,7 @@ static void alc861vd_auto_init(struct hda_codec *codec)
         alc861vd_auto_init_multi_out(codec);
         alc861vd_auto_init_hp_out(codec);
         alc861vd_auto_init_analog_input(codec);
+        alc861vd_auto_init_input_src(codec);
         if (spec->unsol_event)
                 alc_sku_automute(codec);
 }
@@ -14677,6 +14719,8 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
         }
 }
 
+#define alc662_auto_init_input_src      alc882_auto_init_input_src
+
 static int alc662_parse_auto_config(struct hda_codec *codec)
 {
         struct alc_spec *spec = codec->spec;
@@ -14733,6 +14777,7 @@ static void alc662_auto_init(struct hda_codec *codec)
         alc662_auto_init_multi_out(codec);
         alc662_auto_init_hp_out(codec);
         alc662_auto_init_analog_input(codec);
+        alc662_auto_init_input_src(codec);
         if (spec->unsol_event)
                 alc_sku_automute(codec);
 }
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 9a2c16bf94e0..01d7b75f9182 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -36,15 +36,15 @@
  */
 
 /*
- * Xonar DX
- * --------
+ * Xonar D1/DX
+ * -----------
  *
  * CMI8788:
  *
  * I²C <-> CS4398 (front)
  *     <-> CS4362A (surround, center/LFE, back)
  *
- * GPI 0 <- external power present
+ * GPI 0 <- external power present (DX only)
  *
  * GPIO 0 -> enable output to speakers
  * GPIO 1 -> enable front panel I/O
@@ -96,6 +96,7 @@ MODULE_PARM_DESC(enable, "enable card");
 enum {
         MODEL_D2,
         MODEL_D2X,
+        MODEL_D1,
         MODEL_DX,
 };
 
@@ -103,6 +104,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = {
         { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 },
         { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX },
         { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
+        { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 },
         { }
 };
 MODULE_DEVICE_TABLE(pci, xonar_ids);
@@ -313,15 +315,12 @@ static void cs43xx_init(struct oxygen *chip)
         cs4362a_write(chip, 0x01, CS4362A_CPEN);
 }
 
-static void xonar_dx_init(struct oxygen *chip)
+static void xonar_d1_init(struct oxygen *chip)
 {
         struct xonar_data *data = chip->model_data;
 
         data->anti_pop_delay = 800;
         data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
-        data->ext_power_reg = OXYGEN_GPI_DATA;
-        data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
-        data->ext_power_bit = GPI_DX_EXT_POWER;
         data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST;
         data->cs4362a_fm = CS4362A_FM_SINGLE |
                 CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
@@ -345,6 +344,16 @@ static void xonar_dx_init(struct oxygen *chip)
         snd_component_add(chip->card, "CS5361");
 }
 
+static void xonar_dx_init(struct oxygen *chip)
+{
+        struct xonar_data *data = chip->model_data;
+
+        data->ext_power_reg = OXYGEN_GPI_DATA;
+        data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+        data->ext_power_bit = GPI_DX_EXT_POWER;
+        xonar_d1_init(chip);
+}
+
 static void xonar_cleanup(struct oxygen *chip)
 {
         struct xonar_data *data = chip->model_data;
@@ -352,7 +361,7 @@ static void xonar_cleanup(struct oxygen *chip)
         oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
 }
 
-static void xonar_dx_cleanup(struct oxygen *chip)
+static void xonar_d1_cleanup(struct oxygen *chip)
 {
         xonar_cleanup(chip);
         cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
@@ -365,7 +374,7 @@ static void xonar_d2_resume(struct oxygen *chip)
         xonar_enable_output(chip);
 }
 
-static void xonar_dx_resume(struct oxygen *chip)
+static void xonar_d1_resume(struct oxygen *chip)
 {
         cs43xx_init(chip);
         xonar_enable_output(chip);
@@ -513,7 +522,7 @@ static const struct snd_kcontrol_new front_panel_switch = {
         .put = front_panel_put,
 };
 
-static void xonar_dx_ac97_switch(struct oxygen *chip,
+static void xonar_d1_ac97_switch(struct oxygen *chip,
                                  unsigned int reg, unsigned int mute)
 {
         if (reg == AC97_LINE) {
@@ -536,7 +545,7 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
         return 0;
 }
 
-static int xonar_dx_control_filter(struct snd_kcontrol_new *template)
+static int xonar_d1_control_filter(struct snd_kcontrol_new *template)
 {
         if (!strncmp(template->name, "CD Capture ", 11))
                 return 1; /* no CD input */
@@ -548,7 +557,7 @@ static int xonar_mixer_init(struct oxygen *chip)
         return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip));
 }
 
-static int xonar_dx_mixer_init(struct oxygen *chip)
+static int xonar_d1_mixer_init(struct oxygen *chip)
 {
         return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip));
 }
@@ -615,23 +624,51 @@ static const struct oxygen_model xonar_models[] = {
                 .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
                 .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
         },
+        [MODEL_D1] = {
+                .shortname = "Xonar D1",
+                .longname = "Asus Virtuoso 100",
+                .chip = "AV200",
+                .owner = THIS_MODULE,
+                .init = xonar_d1_init,
+                .control_filter = xonar_d1_control_filter,
+                .mixer_init = xonar_d1_mixer_init,
+                .cleanup = xonar_d1_cleanup,
+                .suspend = xonar_d1_cleanup,
+                .resume = xonar_d1_resume,
+                .set_dac_params = set_cs43xx_params,
+                .set_adc_params = set_cs53x1_params,
+                .update_dac_volume = update_cs43xx_volume,
+                .update_dac_mute = update_cs43xx_mute,
+                .ac97_switch = xonar_d1_ac97_switch,
+                .dac_tlv = cs4362a_db_scale,
+                .model_data_size = sizeof(struct xonar_data),
+                .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+                               PLAYBACK_1_TO_SPDIF |
+                               CAPTURE_0_FROM_I2S_2,
+                .dac_channels = 8,
+                .dac_volume_min = 0,
+                .dac_volume_max = 127,
+                .function_flags = OXYGEN_FUNCTION_2WIRE,
+                .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+                .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+        },
         [MODEL_DX] = {
                 .shortname = "Xonar DX",
                 .longname = "Asus Virtuoso 100",
                 .chip = "AV200",
                 .owner = THIS_MODULE,
                 .init = xonar_dx_init,
-                .control_filter = xonar_dx_control_filter,
-                .mixer_init = xonar_dx_mixer_init,
-                .cleanup = xonar_dx_cleanup,
-                .suspend = xonar_dx_cleanup,
-                .resume = xonar_dx_resume,
+                .control_filter = xonar_d1_control_filter,
+                .mixer_init = xonar_d1_mixer_init,
+                .cleanup = xonar_d1_cleanup,
+                .suspend = xonar_d1_cleanup,
+                .resume = xonar_d1_resume,
                 .set_dac_params = set_cs43xx_params,
                 .set_adc_params = set_cs53x1_params,
                 .update_dac_volume = update_cs43xx_volume,
                 .update_dac_mute = update_cs43xx_mute,
                 .gpio_changed = xonar_gpio_changed,
-                .ac97_switch = xonar_dx_ac97_switch,
+                .ac97_switch = xonar_d1_ac97_switch,
                 .dac_tlv = cs4362a_db_scale,
                 .model_data_size = sizeof(struct xonar_data),
                 .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 9ca113326143..54df8baf916f 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -42,7 +42,7 @@
 #include <sound/info.h>
 #include <asm/io.h>
 #include <asm/dma.h>
-#include <asm/dreamcast/sysasic.h>
+#include <mach/sysasic.h>
 #include "aica.h"
 
 MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>");
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
index fee5f8e58957..3f326219f1ec 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -36,8 +36,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/arch/at32ap700x.h>
-#include <asm/arch/portmux.h>
+#include <mach/at32ap700x.h>
+#include <mach/portmux.h>
 
 #include "../codecs/wm8510.h"
 #include "at32-pcm.h"
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
index d47492b2b6e5..7ab48bd25e4c 100644
--- a/sound/soc/at91/at91-pcm.c
+++ b/sound/soc/at91/at91-pcm.c
@@ -28,8 +28,8 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_ssc.h>
+#include <mach/hardware.h>
+#include <mach/at91_ssc.h>
 
 #include "at91-pcm.h"
 
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
index 58d0f00a07b2..e5aada2cb102 100644
--- a/sound/soc/at91/at91-pcm.h
+++ b/sound/soc/at91/at91-pcm.h
@@ -19,7 +19,7 @@
 #ifndef _AT91_PCM_H
 #define _AT91_PCM_H
 
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 
 struct at91_ssc_periph {
         void __iomem    *base;
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index 090e607f8692..5d44515e62e0 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -28,9 +28,9 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_pmc.h>
-#include <asm/arch/at91_ssc.h>
+#include <mach/hardware.h>
+#include <mach/at91_pmc.h>
+#include <mach/at91_ssc.h>
 
 #include "at91-pcm.h"
 #include "at91-ssc.h"
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index d532de954241..b081e83766b7 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -33,8 +33,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/gpio.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
 
 #include "../codecs/wm8731.h"
 #include "at91-pcm.h"
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index e23cb09f0d14..c6a8edf302ad 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -348,8 +348,9 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
         SND_SOC_DAPM_OUTPUT("ROUT1"),
         SND_SOC_DAPM_OUTPUT("LOUT2"),
         SND_SOC_DAPM_OUTPUT("ROUT2"),
-        SND_SOC_DAPM_OUTPUT("MONO"),
+        SND_SOC_DAPM_OUTPUT("MONO1"),
         SND_SOC_DAPM_OUTPUT("OUT3"),
+        SND_SOC_DAPM_OUTPUT("VREF"),
 
         SND_SOC_DAPM_INPUT("LINPUT1"),
         SND_SOC_DAPM_INPUT("LINPUT2"),
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 3ecce5168e94..e44153fa38de 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -82,7 +82,7 @@ static const u16 wm8990_reg[] = {
         0x0003,     /* R35 - ClassD1 */
         0x0000,     /* R36 */
         0x0100,     /* R37 - ClassD3 */
-        0x0000,     /* R38 */
+        0x0079,     /* R38 - ClassD4 */
         0x0000,     /* R39 - Input Mixer1 */
         0x0000,     /* R40 - Input Mixer2 */
         0x0000,     /* R41 - Input Mixer3 */
@@ -311,11 +311,15 @@ SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1,
         WM8990_CDMODE_BIT, 1, 0),
 
 SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME,
-        WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0),
+        WM8990_SPKATTN_SHIFT, WM8990_SPKATTN_MASK, 0),
 SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3,
         WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0),
 SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3,
         WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0),
+SOC_SINGLE_TLV("Speaker Volume", WM8990_CLASSD4,
+        WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("Speaker ZC Switch", WM8990_CLASSD4,
+        WM8990_SPKZC_SHIFT, WM8990_SPKZC_MASK, 0),
 
 SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
         WM8990_LEFT_DAC_DIGITAL_VOLUME,
@@ -920,7 +924,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
         {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
         {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
         {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
-        {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+        {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"},
 
         /* LONMIX */
         {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 6bea57485283..0a08325d5443 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -54,6 +54,7 @@
 #define WM8990_SPEAKER_VOLUME                   0x22
 #define WM8990_CLASSD1                          0x23
 #define WM8990_CLASSD3                          0x25
+#define WM8990_CLASSD4                          0x26
 #define WM8990_INPUT_MIXER1                     0x27
 #define WM8990_INPUT_MIXER2                     0x28
 #define WM8990_INPUT_MIXER3                     0x29
@@ -528,8 +529,8 @@
 /*
  * R34 (0x22) - Speaker Volume
  */
-#define WM8990_SPKVOL_MASK                      0x0003  /* SPKVOL - [1:0] */
-#define WM8990_SPKVOL_SHIFT                     0
+#define WM8990_SPKATTN_MASK                      0x0003  /* SPKATTN - [1:0] */
+#define WM8990_SPKATTN_SHIFT                     0
 
 /*
  * R35 (0x23) - ClassD1
@@ -544,6 +545,15 @@
 #define WM8990_DCGAIN_SHIFT                     3
 #define WM8990_ACGAIN_MASK                      0x0007  /* ACGAIN - [2:0] */
 #define WM8990_ACGAIN_SHIFT                     0
+
+/*
+ * R38 (0x26) - ClassD4
+ */
+#define WM8990_SPKZC_MASK                       0x0001  /* SPKZC */
+#define WM8990_SPKZC_SHIFT                           7  /* SPKZC */
+#define WM8990_SPKVOL_MASK                      0x007F  /* SPKVOL - [6:0] */
+#define WM8990_SPKVOL_SHIFT                          0  /* SPKVOL - [6:0] */
+
 /*
  * R39 (0x27) - Input Mixer1
  */
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 5e2c306399ed..65fdbd81a379 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -19,9 +19,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/mach-types.h>
 #include <asm/dma.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index da2bc5902864..d2d3da9729f2 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -132,12 +132,17 @@ struct fsl_dma_private {
  * Since each link descriptor has a 32-bit byte count field, we set
  * period_bytes_max to the largest 32-bit number.  We also have no maximum
  * number of periods.
+ *
+ * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
+ * limitation in the SSI driver requires the sample rates for playback and
+ * capture to be the same.
  */
 static const struct snd_pcm_hardware fsl_dma_hardware = {
 
         .info                   = SNDRV_PCM_INFO_INTERLEAVED |
                                   SNDRV_PCM_INFO_MMAP |
-                                  SNDRV_PCM_INFO_MMAP_VALID,
+                                  SNDRV_PCM_INFO_MMAP_VALID |
+                                  SNDRV_PCM_INFO_JOINT_DUPLEX,
         .formats                = FSLDMA_PCM_FORMATS,
         .rates                  = FSLDMA_PCM_RATES,
         .rate_min               = 5512,
@@ -322,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
  * fsl_dma_open: open a new substream.
  *
  * Each substream has its own DMA buffer.
+ *
+ * ALSA divides the DMA buffer into N periods.  We create NUM_DMA_LINKS link
+ * descriptors that ping-pong from one period to the next.  For example, if
+ * there are six periods and two link descriptors, this is how they look
+ * before playback starts:
+ *
+ *                 The last link descriptor
+ *   ____________  points back to the first
+ *  |            |
+ *  V            |
+ *  ___    ___   |
+ * |   |->|   |->|
+ * |___|  |___|
+ *   |      |
+ *   |      |
+ *   V      V
+ *  _________________________________________
+ * |      |      |      |      |      |      |  The DMA buffer is
+ * |      |      |      |      |      |      |    divided into 6 parts
+ * |______|______|______|______|______|______|
+ *
+ * and here's how they look after the first period is finished playing:
+ *
+ *   ____________
+ *  |            |
+ *  V            |
+ *  ___    ___   |
+ * |   |->|   |->|
+ * |___|  |___|
+ *   |      |
+ *   |______________
+ *          |       |
+ *          V       V
+ *  _________________________________________
+ * |      |      |      |      |      |      |
+ * |      |      |      |      |      |      |
+ * |______|______|______|______|______|______|
+ *
+ * The first link descriptor now points to the third period.  The DMA
+ * controller is currently playing the second period.  When it finishes, it
+ * will jump back to the first descriptor and play the third period.
+ *
+ * There are four reasons we do this:
+ *
+ * 1. The only way to get the DMA controller to automatically restart the
+ *    transfer when it gets to the end of the buffer is to use chaining
+ *    mode.  Basic direct mode doesn't offer that feature.
+ * 2. We need to receive an interrupt at the end of every period.  The DMA
+ *    controller can generate an interrupt at the end of every link transfer
+ *    (aka segment).  Making each period into a DMA segment will give us the
+ *    interrupts we need.
+ * 3. By creating only two link descriptors, regardless of the number of
+ *    periods, we do not need to reallocate the link descriptors if the
+ *    number of periods changes.
+ * 4. All of the audio data is still stored in a single, contiguous DMA
+ *    buffer, which is what ALSA expects.  We're just dividing it into
+ *    contiguous parts, and creating a link descriptor for each one.
  */
 static int fsl_dma_open(struct snd_pcm_substream *substream)
 {
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct fsl_dma_private *dma_private;
+        struct ccsr_dma_channel __iomem *dma_channel;
         dma_addr_t ld_buf_phys;
+        u64 temp_link;          /* Pointer to next link descriptor */
+        u32 mr;
         unsigned int channel;
         int ret = 0;
+        unsigned int i;
 
         /*
          * Reject any DMA buffer whose size is not a multiple of the period
@@ -390,68 +456,74 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
         snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
         runtime->private_data = dma_private;
 
+        /* Program the fixed DMA controller parameters */
+
+        dma_channel = dma_private->dma_channel;
+
+        temp_link = dma_private->ld_buf_phys +
+                sizeof(struct fsl_dma_link_descriptor);
+
+        for (i = 0; i < NUM_DMA_LINKS; i++) {
+                struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+                link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+                link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+                link->next = cpu_to_be64(temp_link);
+
+                temp_link += sizeof(struct fsl_dma_link_descriptor);
+        }
+        /* The last link descriptor points to the first */
+        dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
+
+        /* Tell the DMA controller where the first link descriptor is */
+        out_be32(&dma_channel->clndar,
+                CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
+        out_be32(&dma_channel->eclndar,
+                CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
+
+        /* The manual says the BCR must be clear before enabling EMP */
+        out_be32(&dma_channel->bcr, 0);
+
+        /*
+         * Program the mode register for interrupts, external master control,
+         * and source/destination hold.  Also clear the Channel Abort bit.
+         */
+        mr = in_be32(&dma_channel->mr) &
+                ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
+
+        /*
+         * We want External Master Start and External Master Pause enabled,
+         * because the SSI is controlling the DMA controller.  We want the DMA
+         * controller to be set up in advance, and then we signal only the SSI
+         * to start transferring.
+         *
+         * We want End-Of-Segment Interrupts enabled, because this will generate
+         * an interrupt at the end of each segment (each link descriptor
+         * represents one segment).  Each DMA segment is the same thing as an
+         * ALSA period, so this is how we get an interrupt at the end of every
+         * period.
+         *
+         * We want Error Interrupt enabled, so that we can get an error if
+         * the DMA controller is mis-programmed somehow.
+         */
+        mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
+                CCSR_DMA_MR_EMS_EN;
+
+        /* For playback, we want the destination address to be held.  For
+           capture, set the source address to be held. */
+        mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+                CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
+
+        out_be32(&dma_channel->mr, mr);
+
         return 0;
 }
 
 /**
- * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors.
+ * fsl_dma_hw_params: continue initializing the DMA links
  *
- * ALSA divides the DMA buffer into N periods.  We create NUM_DMA_LINKS link
- * descriptors that ping-pong from one period to the next.  For example, if
- * there are six periods and two link descriptors, this is how they look
- * before playback starts:
- *
- *                 The last link descriptor
- *   ____________  points back to the first
- *  |            |
- *  V            |
- *  ___    ___   |
- * |   |->|   |->|
- * |___|  |___|
- *   |      |
- *   |      |
- *   V      V
- *  _________________________________________
- * |      |      |      |      |      |      |  The DMA buffer is
- * |      |      |      |      |      |      |    divided into 6 parts
- * |______|______|______|______|______|______|
- *
- * and here's how they look after the first period is finished playing:
- *
- *   ____________
- *  |            |
- *  V            |
- *  ___    ___   |
- * |   |->|   |->|
- * |___|  |___|
- *   |      |
- *   |______________
- *          |       |
- *          V       V
- *  _________________________________________
- * |      |      |      |      |      |      |
- * |      |      |      |      |      |      |
- * |______|______|______|______|______|______|
- *
- * The first link descriptor now points to the third period.  The DMA
- * controller is currently playing the second period.  When it finishes, it
- * will jump back to the first descriptor and play the third period.
- *
- * There are four reasons we do this:
- *
- * 1. The only way to get the DMA controller to automatically restart the
- *    transfer when it gets to the end of the buffer is to use chaining
- *    mode.  Basic direct mode doesn't offer that feature.
- * 2. We need to receive an interrupt at the end of every period.  The DMA
- *    controller can generate an interrupt at the end of every link transfer
- *    (aka segment).  Making each period into a DMA segment will give us the
- *    interrupts we need.
- * 3. By creating only two link descriptors, regardless of the number of
- *    periods, we do not need to reallocate the link descriptors if the
- *    number of periods changes.
- * 4. All of the audio data is still stored in a single, contiguous DMA
- *    buffer, which is what ALSA expects.  We're just dividing it into
- *    contiguous parts, and creating a link descriptor for each one.
+ * This function obtains hardware parameters about the opened stream and
+ * programs the DMA controller accordingly.
  *
  * Note that due to a quirk of the SSI's STX register, the target address
  * for the DMA operations depends on the sample size.  So we don't program
@@ -463,11 +535,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 {
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct fsl_dma_private *dma_private = runtime->private_data;
-        struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
 
         dma_addr_t temp_addr;   /* Pointer to next period */
-        u64 temp_link;          /* Pointer to next link descriptor */
-        u32 mr;                 /* Temporary variable for MR register */
 
         unsigned int i;
 
@@ -485,8 +554,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
                 dma_private->dma_buf_next = dma_private->dma_buf_phys;
 
         /*
-         * Initialize each link descriptor.
-         *
          * The actual address in STX0 (destination for playback, source for
          * capture) is based on the sample size, but we don't know the sample
          * size in this function, so we'll have to adjust that later.  See
@@ -502,16 +569,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
          * buffer itself.
          */
         temp_addr = substream->dma_buffer.addr;
-        temp_link = dma_private->ld_buf_phys +
-                sizeof(struct fsl_dma_link_descriptor);
 
         for (i = 0; i < NUM_DMA_LINKS; i++) {
                 struct fsl_dma_link_descriptor *link = &dma_private->link[i];
 
                 link->count = cpu_to_be32(period_size);
-                link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-                link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-                link->next = cpu_to_be64(temp_link);
 
                 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
                         link->source_addr = cpu_to_be32(temp_addr);
@@ -519,51 +581,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
                         link->dest_addr = cpu_to_be32(temp_addr);
 
                 temp_addr += period_size;
-                temp_link += sizeof(struct fsl_dma_link_descriptor);
         }
-        /* The last link descriptor points to the first */
-        dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
-
-        /* Tell the DMA controller where the first link descriptor is */
-        out_be32(&dma_channel->clndar,
-                CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
-        out_be32(&dma_channel->eclndar,
-                CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
-
-        /* The manual says the BCR must be clear before enabling EMP */
-        out_be32(&dma_channel->bcr, 0);
-
-        /*
-         * Program the mode register for interrupts, external master control,
-         * and source/destination hold.  Also clear the Channel Abort bit.
-         */
-        mr = in_be32(&dma_channel->mr) &
-                ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
-
-        /*
-         * We want External Master Start and External Master Pause enabled,
-         * because the SSI is controlling the DMA controller.  We want the DMA
-         * controller to be set up in advance, and then we signal only the SSI
-         * to start transfering.
-         *
-         * We want End-Of-Segment Interrupts enabled, because this will generate
-         * an interrupt at the end of each segment (each link descriptor
-         * represents one segment).  Each DMA segment is the same thing as an
-         * ALSA period, so this is how we get an interrupt at the end of every
-         * period.
-         *
-         * We want Error Interrupt enabled, so that we can get an error if
-         * the DMA controller is mis-programmed somehow.
-         */
-        mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
-                CCSR_DMA_MR_EMS_EN;
-
-        /* For playback, we want the destination address to be held.  For
-           capture, set the source address to be held. */
-        mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-                CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
-
-        out_be32(&dma_channel->mr, mr);
 
         return 0;
 }
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 71bff33f5528..157a7895ffa1 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -67,6 +67,8 @@
  * @ssi: pointer to the SSI's registers
  * @ssi_phys: physical address of the SSI registers
  * @irq: IRQ of this SSI
+ * @first_stream: pointer to the stream that was opened first
+ * @second_stream: pointer to second stream
  * @dev: struct device pointer
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
@@ -79,6 +81,8 @@ struct fsl_ssi_private {
         struct ccsr_ssi __iomem *ssi;
         dma_addr_t ssi_phys;
         unsigned int irq;
+        struct snd_pcm_substream *first_stream;
+        struct snd_pcm_substream *second_stream;
         struct device *dev;
         unsigned int playback;
         unsigned int capture;
@@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
                  */
         }
 
+        if (!ssi_private->first_stream)
+                ssi_private->first_stream = substream;
+        else {
+                /* This is the second stream open, so we need to impose sample
+                 * rate and maybe sample size constraints.  Note that this can
+                 * cause a race condition if the second stream is opened before
+                 * the first stream is fully initialized.
+                 *
+                 * We provide some protection by checking to make sure the first
+                 * stream is initialized, but it's not perfect.  ALSA sometimes
+                 * re-initializes the driver with a different sample rate or
+                 * size.  If the second stream is opened before the first stream
+                 * has received its final parameters, then the second stream may
+                 * be constrained to the wrong sample rate or size.
+                 *
+                 * FIXME: This code does not handle opening and closing streams
+                 * repeatedly.  If you open two streams and then close the first
+                 * one, you may not be able to open another stream until you
+                 * close the second one as well.
+                 */
+                struct snd_pcm_runtime *first_runtime =
+                        ssi_private->first_stream->runtime;
+
+                if (!first_runtime->rate || !first_runtime->sample_bits) {
+                        dev_err(substream->pcm->card->dev,
+                                "set sample rate and size in %s stream first\n",
+                                substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+                                ? "capture" : "playback");
+                        return -EAGAIN;
+                }
+
+                snd_pcm_hw_constraint_minmax(substream->runtime,
+                        SNDRV_PCM_HW_PARAM_RATE,
+                        first_runtime->rate, first_runtime->rate);
+
+                snd_pcm_hw_constraint_minmax(substream->runtime,
+                        SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+                        first_runtime->sample_bits,
+                        first_runtime->sample_bits);
+
+                ssi_private->second_stream = substream;
+        }
+
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
                 ssi_private->playback++;
 
@@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
         struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
 
         struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-        u32 wl;
 
-        wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
+        if (substream == ssi_private->first_stream) {
+                u32 wl;
 
-        clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+                /* The SSI should always be disabled at this points (SSIEN=0) */
+                wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
 
-        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                /* In synchronous mode, the SSI uses STCCR for capture */
                 clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-        else
-                clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-
-        setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+        }
 
         return 0;
 }
@@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
         case SNDRV_PCM_TRIGGER_RESUME:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
                 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-                        setbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+                        clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+                        setbits32(&ssi->scr,
+                                CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
                 } else {
-                        setbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+                        clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+                        setbits32(&ssi->scr,
+                                CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
 
                         /*
                          * I think we need this delay to allow time for the SSI
@@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
         if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
                 ssi_private->capture--;
 
+        if (ssi_private->first_stream == substream)
+                ssi_private->first_stream = ssi_private->second_stream;
+
+        ssi_private->second_stream = NULL;
+
         /*
          * If this is the last active substream, disable the SSI and release
          * the IRQ.
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 02cec96859b8..7694621ec40b 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -29,9 +29,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 #include <linux/gpio.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/mcbsp.h>
 
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 00b0c9d73cd4..35310e16d7f3 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -30,9 +30,9 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/arch/control.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/control.h>
+#include <mach/dma.h>
+#include <mach/mcbsp.h>
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e092f3d836d0..690bfeaec4a0 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -27,7 +27,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 #include "omap-pcm.h"
 
 static const struct snd_pcm_hardware omap_pcm_hardware = {
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index c0294464a23a..0a53f72077fd 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -25,10 +25,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/corgi.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/corgi.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 06e8afb25277..6781c5be242f 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -21,9 +21,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 02dcac39cdf6..d9c3f7b28be2 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -30,9 +30,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 65a4e9a8c39e..a4697f7e2921 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -26,10 +26,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/locomo.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/poodle.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/poodle.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
 #include "pxa2xx-pcm.h"
@@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on poodle */
-static int poodle_shutdown(struct snd_pcm_substream *substream)
+static void poodle_shutdown(struct snd_pcm_substream *substream)
 {
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct snd_soc_codec *codec = rtd->socdev->codec;
-
         /* set = unmute headphone */
         locomo_gpio_write(&poodle_locomo_device.dev,
                 POODLE_LOCOMO_GPIO_MUTE_L, 1);
         locomo_gpio_write(&poodle_locomo_device.dev,
                 POODLE_LOCOMO_GPIO_MUTE_R, 1);
-        return 0;
 }
 
 static int poodle_hw_params(struct snd_pcm_substream *substream,
@@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = {
         SOC_ENUM_SINGLE_EXT(2, spk_function),
 };
 
-static const snd_kcontrol_new_t wm8731_poodle_controls[] = {
+static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
         SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
                 poodle_set_jack),
         SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 059af815ea0c..d94a495bd6bd 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -26,10 +26,10 @@
 
 #include <asm/irq.h>
 #include <linux/mutex.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 8f96d87f7b4b..8548818eea08 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -21,10 +21,10 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2df03ee5819e..4345f387fe41 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -22,9 +22,9 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 64385797da5d..37cb768fc933 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -26,10 +26,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/akita.h>
-#include <asm/arch/spitz.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/akita.h>
+#include <mach/spitz.h>
 #include "../codecs/wm8750.h"
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
@@ -297,7 +297,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
         snd_soc_dapm_disable_pin(codec, "LINPUT3");
         snd_soc_dapm_disable_pin(codec, "RINPUT3");
         snd_soc_dapm_disable_pin(codec, "OUT3");
-        snd_soc_dapm_disable_pin(codec, "MONO");
+        snd_soc_dapm_disable_pin(codec, "MONO1");
 
         /* Add spitz specific controls */
         for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index fe6cca9c9e76..2baaa750f123 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -29,11 +29,10 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/tosa.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/tosa.h>
+#include <mach/tosa.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 4d7a9aa15f1a..8089f8ee05c0 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -24,14 +24,13 @@
 #include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 
-#include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-clock.h>
+#include <mach/regs-gpio.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 #include <linux/io.h>
-#include <asm/arch/spi-gpio.h>
+#include <mach/spi-gpio.h>
 
 #include <asm/plat-s3c24xx/regs-iis.h>
 
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ee4676ed1283..ded7d995a922 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -28,16 +28,16 @@
 #include <sound/pcm_params.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 
 #include <linux/io.h>
 #include <asm/dma.h>
 
 #include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
 
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/dma.h>
+#include <mach/regs-gpio.h>
+#include <mach/audio.h>
+#include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c2412-i2s.h"
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 783349b7fede..19c5c3cf5d8c 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -27,13 +27,13 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/plat-s3c/regs-ac97.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
 #include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 397524282b57..ba4476b55fbc 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -27,12 +27,12 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
 #include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 
 #include <asm/plat-s3c24xx/regs-iis.h>
 
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index cef79b34dc6f..e13e614bada9 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -27,9 +27,9 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/dma.h>
+#include <mach/audio.h>
 
 #include "s3c24xx-pcm.h"
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 820347c9ae4b..f9d100bc8479 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 
         return 0;
 }
+EXPORT_SYMBOL_GPL(dapm_reg_event);
 
 /*
  * Scan each dapm widget for complete audio path.