Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: emu10k1: allow high-resolution mixer controls
  ALSA: pcm: fix delta calculation at boundary wraparound
  ALSA: hda_intel: fix handling of non-completion stream interrupts
  ALSA: usb/caiaq: fix Traktor Kontrol X1 ABS_HAT2X axis
  ALSA: hda: Fix model quirk for Dell M1730
  ALSA: hda - iMac9,1 sound fixes
  ALSA: hda: Use LPIB for Toshiba A100-259
  ALSA: hda: Use LPIB for Acer Aspire 5110
  ALSA: aw2-alsa.c: use pci_ids.h defines and fix checkpatch.pl noise
  ALSA: usb-audio: add support for Akai MPD16
  ALSA: pcm: fix the fix of the runtime->boundary calculation

13 files changed, 219 insertions, 102 deletions

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a2ff86189d2a..e9d98be190c5 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -345,7 +345,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
                 new_hw_ptr = hw_base + pos;
         }
       __delta:
-        delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary;
+        delta = new_hw_ptr - old_hw_ptr;
+        if (delta < 0)
+                delta += runtime->boundary;
         if (xrun_debug(substream, in_interrupt ?
                         XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) {
                 char name[16];
@@ -439,8 +441,13 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
                 snd_pcm_playback_silence(substream, new_hw_ptr);
 
         if (in_interrupt) {
-                runtime->hw_ptr_interrupt = new_hw_ptr -
-                                (new_hw_ptr % runtime->period_size);
+                delta = new_hw_ptr - runtime->hw_ptr_interrupt;
+                if (delta < 0)
+                        delta += runtime->boundary;
+                delta -= (snd_pcm_uframes_t)delta % runtime->period_size;
+                runtime->hw_ptr_interrupt += delta;
+                if (runtime->hw_ptr_interrupt >= runtime->boundary)
+                        runtime->hw_ptr_interrupt -= runtime->boundary;
         }
         runtime->hw_ptr_base = hw_base;
         runtime->status->hw_ptr = new_hw_ptr;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 644c2bb17b86..303ac04ff6e4 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -27,7 +27,6 @@
 #include <linux/pm_qos_params.h>
 #include <linux/uio.h>
 #include <linux/dma-mapping.h>
-#include <linux/math64.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/info.h>
@@ -370,38 +369,6 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime)
         return usecs;
 }
 
-static int calc_boundary(struct snd_pcm_runtime *runtime)
-{
-        u_int64_t boundary;
-
-        boundary = (u_int64_t)runtime->buffer_size *
-                   (u_int64_t)runtime->period_size;
-#if BITS_PER_LONG < 64
-        /* try to find lowest common multiple for buffer and period */
-        if (boundary > LONG_MAX - runtime->buffer_size) {
-                u_int32_t remainder = -1;
-                u_int32_t divident = runtime->buffer_size;
-                u_int32_t divisor = runtime->period_size;
-                while (remainder) {
-                        remainder = divident % divisor;
-                        if (remainder) {
-                                divident = divisor;
-                                divisor = remainder;
-                        }
-                }
-                boundary = div_u64(boundary, divisor);
-                if (boundary > LONG_MAX - runtime->buffer_size)
-                        return -ERANGE;
-        }
-#endif
-        if (boundary == 0)
-                return -ERANGE;
-        runtime->boundary = boundary;
-        while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
-                runtime->boundary *= 2;
-        return 0;
-}
-
 static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
                              struct snd_pcm_hw_params *params)
 {
@@ -477,9 +444,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
         runtime->stop_threshold = runtime->buffer_size;
         runtime->silence_threshold = 0;
         runtime->silence_size = 0;
-        err = calc_boundary(runtime);
-        if (err < 0)
-                goto _error;
+        runtime->boundary = runtime->buffer_size;
+        while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
+                runtime->boundary *= 2;
 
         snd_pcm_timer_resolution_change(substream);
         runtime->status->state = SNDRV_PCM_STATE_SETUP;
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 67921f93a41e..c15002242d98 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -26,7 +26,7 @@
 #include <linux/slab.h>
 #include <linux/interrupt.h>
 #include <linux/delay.h>
-#include <asm/io.h>
+#include <linux/io.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/pcm.h>
@@ -44,9 +44,6 @@ MODULE_LICENSE("GPL");
 /*********************************
  * DEFINES
  ********************************/
-#define PCI_VENDOR_ID_SAA7146             0x1131
-#define PCI_DEVICE_ID_SAA7146             0x7146
-
 #define CTL_ROUTE_ANALOG 0
 #define CTL_ROUTE_DIGITAL 1
 
@@ -165,7 +162,7 @@ module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
 
 static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = {
-        {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
+        {PCI_VENDOR_ID_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, 0, 0,
          0, 0, 0},
         {0}
 };
@@ -419,7 +416,7 @@ static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream)
 {
         struct snd_pcm_runtime *runtime = substream->runtime;
 
-        snd_printdd(KERN_DEBUG "aw2: Playback_open \n");
+        snd_printdd(KERN_DEBUG "aw2: Playback_open\n");
         runtime->hw = snd_aw2_playback_hw;
         return 0;
 }
@@ -435,7 +432,7 @@ static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream)
 {
         struct snd_pcm_runtime *runtime = substream->runtime;
 
-        snd_printdd(KERN_DEBUG "aw2: Capture_open \n");
+        snd_printdd(KERN_DEBUG "aw2: Capture_open\n");
         runtime->hw = snd_aw2_capture_hw;
         return 0;
 }
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 4b302d86f5f2..7a9401462c1c 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -35,6 +35,7 @@
 #include <linux/vmalloc.h>
 #include <linux/init.h>
 #include <linux/mutex.h>
+#include <linux/moduleparam.h>
 
 #include <sound/core.h>
 #include <sound/tlv.h>
@@ -50,6 +51,10 @@
 #define EMU10K1_CENTER_LFE_FROM_FRONT
 #endif
 
+static bool high_res_gpr_volume;
+module_param(high_res_gpr_volume, bool, 0444);
+MODULE_PARM_DESC(high_res_gpr_volume, "GPR mixer controls use 31-bit range.");
+
 /*
  *  Tables
  */ 
@@ -296,6 +301,7 @@ static const u32 db_table[101] = {
 
 /* EMU10k1/EMU10k2 DSP control db gain */
 static const DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1);
+static const DECLARE_TLV_DB_LINEAR(snd_emu10k1_db_linear, TLV_DB_GAIN_MUTE, 0);
 
 static const u32 onoff_table[2] = {
         0x00000000, 0x00000001
@@ -1072,10 +1078,17 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
         strcpy(ctl->id.name, name);
         ctl->vcount = ctl->count = 1;
         ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
-        ctl->min = 0;
-        ctl->max = 100;
-        ctl->tlv = snd_emu10k1_db_scale1;
-        ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;    
+        if (high_res_gpr_volume) {
+                ctl->min = 0;
+                ctl->max = 0x7fffffff;
+                ctl->tlv = snd_emu10k1_db_linear;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_NONE;
+        } else {
+                ctl->min = 0;
+                ctl->max = 100;
+                ctl->tlv = snd_emu10k1_db_scale1;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+        }
 }
 
 static void __devinit
@@ -1087,10 +1100,17 @@ snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
         ctl->vcount = ctl->count = 2;
         ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
         ctl->gpr[1] = gpr + 1; ctl->value[1] = defval;
-        ctl->min = 0;
-        ctl->max = 100;
-        ctl->tlv = snd_emu10k1_db_scale1;
-        ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+        if (high_res_gpr_volume) {
+                ctl->min = 0;
+                ctl->max = 0x7fffffff;
+                ctl->tlv = snd_emu10k1_db_linear;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_NONE;
+        } else {
+                ctl->min = 0;
+                ctl->max = 100;
+                ctl->tlv = snd_emu10k1_db_scale1;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+        }
 }
 
 static void __devinit
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 170610e1d7da..77e22c2a8caa 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1097,6 +1097,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
         struct azx *chip = dev_id;
         struct azx_dev *azx_dev;
         u32 status;
+        u8 sd_status;
         int i, ok;
 
         spin_lock(&chip->reg_lock);
@@ -1110,8 +1111,10 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
         for (i = 0; i < chip->num_streams; i++) {
                 azx_dev = &chip->azx_dev[i];
                 if (status & azx_dev->sd_int_sta_mask) {
+                        sd_status = azx_sd_readb(azx_dev, SD_STS);
                         azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
-                        if (!azx_dev->substream || !azx_dev->running)
+                        if (!azx_dev->substream || !azx_dev->running ||
+                            !(sd_status & SD_INT_COMPLETE))
                                 continue;
                         /* check whether this IRQ is really acceptable */
                         ok = azx_position_ok(chip, azx_dev);
@@ -2279,12 +2282,14 @@ static int azx_dev_free(struct snd_device *device)
  * white/black-listing for position_fix
  */
 static struct snd_pci_quirk position_fix_list[] __devinitdata = {
+        SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
-        SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 53538b0f9991..17d4548cc353 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7025,6 +7025,14 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
         },
 };
 
+static struct hda_input_mux alc889A_imac91_capture_source = {
+        .num_items = 2,
+        .items = {
+                { "Mic", 0x01 },
+                { "Line", 0x2 }, /* Not sure! */
+        },
+};
+
 /*
  * 2ch mode
  */
@@ -7486,15 +7494,8 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = {
 };
 
 static struct snd_kcontrol_new alc885_imac91_mixer[] = {
-        HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
-        HDA_BIND_MUTE   ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT),
-        HDA_CODEC_MUTE  ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT),
-        HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
-        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-        HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
-        HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
+        HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+        HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
         { } /* end */
 };
 
@@ -7995,61 +7996,56 @@ static struct hda_verb alc885_mbp3_init_verbs[] = {
 
 /* iMac 9,1 */
 static struct hda_verb alc885_imac91_init_verbs[] = {
-        /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
-        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-        /* Rear mixer */
-        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-        /* HP Pin: output 0 (0x0c) */
+        /* Internal Speaker Pin (0x0c) */
+        {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* HP Pin: Rear */
         {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
         {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
         {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-        /* Internal Speakers: output 0 (0x0d) */
-        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)},
+        /* Line in Rear */
+        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
         {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-        {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-        /* Mic (rear) pin: input vref at 80% */
-        {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
         /* Front Mic pin: input vref at 80% */
         {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
         {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-        /* Line In pin: use output 1 when in LineOut mode */
-        {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-        {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-
-        /* FIXME: use matrix-type input source selection */
-        /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
-        /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+        /* Rear mixer */
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-        /* Input mixer2 */
+        /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-        /* Input mixer3 */
+        /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-        /* ADC1: mute amp left and right */
+        /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
         {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-        /* ADC2: mute amp left and right */
+        /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
         {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-        /* ADC3: mute amp left and right */
+        /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
         {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
         { }
 };
 
@@ -8118,7 +8114,7 @@ static void alc885_imac91_setup(struct hda_codec *codec)
         struct alc_spec *spec = codec->spec;
 
         spec->autocfg.hp_pins[0] = 0x14;
-        spec->autocfg.speaker_pins[0] = 0x15;
+        spec->autocfg.speaker_pins[0] = 0x18;
         spec->autocfg.speaker_pins[1] = 0x1a;
 }
 
@@ -9627,14 +9623,14 @@ static struct alc_config_preset alc882_presets[] = {
                 .init_hook = alc885_imac24_init_hook,
         },
         [ALC885_IMAC91] = {
-                .mixers = { alc885_imac91_mixer, alc882_chmode_mixer },
+                .mixers = {alc885_imac91_mixer},
                 .init_verbs = { alc885_imac91_init_verbs,
                                 alc880_gpio1_init_verbs },
                 .num_dacs = ARRAY_SIZE(alc882_dac_nids),
                 .dac_nids = alc882_dac_nids,
-                .channel_mode = alc885_mbp_4ch_modes,
-                .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
-                .input_mux = &alc882_capture_source,
+                .channel_mode = alc885_mba21_ch_modes,
+                .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
+                .input_mux = &alc889A_imac91_capture_source,
                 .dig_out_nid = ALC882_DIGOUT_NID,
                 .dig_in_nid = ALC882_DIGIN_NID,
                 .unsol_event = alc_automute_amp_unsol_event,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a0e06d82da1f..f1e7babd6920 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2078,12 +2078,12 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
         SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000,
                            "Intel D965", STAC_D965_3ST),
         /* Dell 3 stack systems */
-        SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_3ST),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01dd, "Dell Dimension E520", STAC_DELL_3ST),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01ed, "Dell     ", STAC_DELL_3ST),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f4, "Dell     ", STAC_DELL_3ST),
         /* Dell 3 stack systems with verb table in BIOS */
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
+        SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_BIOS),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0227, "Dell Vostro 1400  ", STAC_DELL_BIOS),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022e, "Dell     ", STAC_DELL_BIOS),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell Inspiron 1525", STAC_DELL_BIOS),
diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 8bbfbfd4c658..dcb620796d9e 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -171,7 +171,7 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev,
                 input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8)  | buf[5]);
                 input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]);
                 input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8)  | buf[3]);
-                input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 8) | buf[15]);
+                input_report_abs(input_dev, ABS_HAT2X, (buf[14] << 8) | buf[15]);
                 input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8)  | buf[1]);
                 input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]);
                 input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8)  | buf[7]);
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 8b1e4b124a9f..46785643c66d 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -645,6 +645,105 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = {
 };
 
 /*
+ * AKAI MPD16 protocol:
+ *
+ * For control port (endpoint 1):
+ * ==============================
+ * One or more chunks consisting of first byte of (0x10 | msg_len) and then a
+ * SysEx message (msg_len=9 bytes long).
+ *
+ * For data port (endpoint 2):
+ * ===========================
+ * One or more chunks consisting of first byte of (0x20 | msg_len) and then a
+ * MIDI message (msg_len bytes long)
+ *
+ * Messages sent: Active Sense, Note On, Poly Pressure, Control Change.
+ */
+static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep,
+                                   uint8_t *buffer, int buffer_length)
+{
+        unsigned int pos = 0;
+        unsigned int len = (unsigned int)buffer_length;
+        while (pos < len) {
+                unsigned int port = (buffer[pos] >> 4) - 1;
+                unsigned int msg_len = buffer[pos] & 0x0f;
+                pos++;
+                if (pos + msg_len <= len && port < 2)
+                        snd_usbmidi_input_data(ep, 0, &buffer[pos], msg_len);
+                pos += msg_len;
+        }
+}
+
+#define MAX_AKAI_SYSEX_LEN 9
+
+static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep,
+                                    struct urb *urb)
+{
+        uint8_t *msg;
+        int pos, end, count, buf_end;
+        uint8_t tmp[MAX_AKAI_SYSEX_LEN];
+        struct snd_rawmidi_substream *substream = ep->ports[0].substream;
+
+        if (!ep->ports[0].active)
+                return;
+
+        msg = urb->transfer_buffer + urb->transfer_buffer_length;
+        buf_end = ep->max_transfer - MAX_AKAI_SYSEX_LEN - 1;
+
+        /* only try adding more data when there's space for at least 1 SysEx */
+        while (urb->transfer_buffer_length < buf_end) {
+                count = snd_rawmidi_transmit_peek(substream,
+                                                  tmp, MAX_AKAI_SYSEX_LEN);
+                if (!count) {
+                        ep->ports[0].active = 0;
+                        return;
+                }
+                /* try to skip non-SysEx data */
+                for (pos = 0; pos < count && tmp[pos] != 0xF0; pos++)
+                        ;
+
+                if (pos > 0) {
+                        snd_rawmidi_transmit_ack(substream, pos);
+                        continue;
+                }
+
+                /* look for the start or end marker */
+                for (end = 1; end < count && tmp[end] < 0xF0; end++)
+                        ;
+
+                /* next SysEx started before the end of current one */
+                if (end < count && tmp[end] == 0xF0) {
+                        /* it's incomplete - drop it */
+                        snd_rawmidi_transmit_ack(substream, end);
+                        continue;
+                }
+                /* SysEx complete */
+                if (end < count && tmp[end] == 0xF7) {
+                        /* queue it, ack it, and get the next one */
+                        count = end + 1;
+                        msg[0] = 0x10 | count;
+                        memcpy(&msg[1], tmp, count);
+                        snd_rawmidi_transmit_ack(substream, count);
+                        urb->transfer_buffer_length += count + 1;
+                        msg += count + 1;
+                        continue;
+                }
+                /* less than 9 bytes and no end byte - wait for more */
+                if (count < MAX_AKAI_SYSEX_LEN) {
+                        ep->ports[0].active = 0;
+                        return;
+                }
+                /* 9 bytes and no end marker in sight - malformed, skip it */
+                snd_rawmidi_transmit_ack(substream, count);
+        }
+}
+
+static struct usb_protocol_ops snd_usbmidi_akai_ops = {
+        .input = snd_usbmidi_akai_input,
+        .output = snd_usbmidi_akai_output,
+};
+
+/*
  * Novation USB MIDI protocol: number of data bytes is in the first byte
  * (when receiving) (+1!) or in the second byte (when sending); data begins
  * at the third byte.
@@ -1434,6 +1533,11 @@ static struct port_info {
         EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"),
         EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"),
         EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"),
+        /* Akai MPD16 */
+        CONTROL_PORT(0x09e8, 0x0062, 0, "%s Control"),
+        PORT_INFO(0x09e8, 0x0062, 1, "%s MIDI", 0,
+                SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |
+                SNDRV_SEQ_PORT_TYPE_HARDWARE),
         /* Access Music Virus TI */
         EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"),
         PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0,
@@ -2035,6 +2139,12 @@ int snd_usbmidi_create(struct snd_card *card,
                 umidi->usb_protocol_ops = &snd_usbmidi_cme_ops;
                 err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
                 break;
+        case QUIRK_MIDI_AKAI:
+                umidi->usb_protocol_ops = &snd_usbmidi_akai_ops;
+                err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+                /* endpoint 1 is input-only */
+                endpoints[1].out_cables = 0;
+                break;
         default:
                 snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
                 err = -ENXIO;
diff --git a/sound/usb/midi.h b/sound/usb/midi.h
index 2089ec987c66..2fca80b744c0 100644
--- a/sound/usb/midi.h
+++ b/sound/usb/midi.h
@@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info {
 
 /* for QUIRK_MIDI_CME, data is NULL */
 
+/* for QUIRK_MIDI_AKAI, data is NULL */
+
 int snd_usbmidi_create(struct snd_card *card,
                        struct usb_interface *iface,
                        struct list_head *midi_list,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 91ddef31bcbd..f8797f61a24b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
         }
 },
 
+/* AKAI devices */
+{
+        USB_DEVICE(0x09e8, 0x0062),
+        .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+                .vendor_name = "AKAI",
+                .product_name = "MPD16",
+                .ifnum = 0,
+                .type = QUIRK_MIDI_AKAI,
+        }
+},
+
 /* TerraTec devices */
 {
         USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 136e5b4cf6de..b45e54c09ba2 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
                 [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk,
                 [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
                 [QUIRK_MIDI_CME] = create_any_midi_quirk,
+                [QUIRK_MIDI_AKAI] = create_any_midi_quirk,
                 [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
                 [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
                 [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index d679e72a3e5c..06ebf24d3a4d 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -74,6 +74,7 @@ enum quirk_type {
         QUIRK_MIDI_FASTLANE,
         QUIRK_MIDI_EMAGIC,
         QUIRK_MIDI_CME,
+        QUIRK_MIDI_AKAI,
         QUIRK_MIDI_US122L,
         QUIRK_AUDIO_STANDARD_INTERFACE,
         QUIRK_AUDIO_FIXED_ENDPOINT,