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

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;

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;

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;
 }

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

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),

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,

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),

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]);

sound/usb/midi.c

@@ -644,6 +644,105 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = {
 	.output_packet = snd_usbmidi_output_standard_packet,
 };
 
+/*
+ * 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
@@ -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;

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,

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),

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,

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,