Merge branch 'x86/apic' into irq/numa

Merge reason: both topics modify the APIC code but were able to do it in
              parallel so far. An upcoming patch generates a conflict so
              merge them to avoid the conflict.

Signed-off-by: Ingo Molnar <mingo@elte.hu>

21 files changed, 105 insertions, 30 deletions

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 63d088f2265f..a2a792c18c40 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,12 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 			new_hw_ptr = hw_base + pos;
 		}
 	}
+	/* Skip the jiffies check for hardwares with BATCH flag.
+	 * Such hardware usually just increases the position at each IRQ,
+	 * thus it can't give any strange position.
+	 */
+	if (runtime->hw.info & SNDRV_PCM_INFO_BATCH)
+		goto no_jiffies_check;
 	hdelta = new_hw_ptr - old_hw_ptr;
 	jdelta = jiffies - runtime->hw_ptr_jiffies;
 	if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
@@ -272,6 +278,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 		hw_base -= hw_base % runtime->buffer_size;
 		delta = 0;
 	}
+ no_jiffies_check:
 	if (delta > runtime->period_size + runtime->period_size / 2) {
 		hw_ptr_error(substream,
 			     "Lost interrupts? "
diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
index 906454413ed2..3a1526ae1729 100644
--- a/sound/isa/msnd/msnd.c
+++ b/sound/isa/msnd/msnd.c
@@ -438,7 +438,8 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip,
 static struct snd_pcm_hardware snd_msnd_playback = {
 	.info =			SNDRV_PCM_INFO_MMAP |
 				SNDRV_PCM_INFO_INTERLEAVED |
-				SNDRV_PCM_INFO_MMAP_VALID,
+				SNDRV_PCM_INFO_MMAP_VALID |
+				SNDRV_PCM_INFO_BATCH,
 	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =		SNDRV_PCM_RATE_8000_48000,
 	.rate_min =		8000,
@@ -456,7 +457,8 @@ static struct snd_pcm_hardware snd_msnd_playback = {
 static struct snd_pcm_hardware snd_msnd_capture = {
 	.info =			SNDRV_PCM_INFO_MMAP |
 				SNDRV_PCM_INFO_INTERLEAVED |
-				SNDRV_PCM_INFO_MMAP_VALID,
+				SNDRV_PCM_INFO_MMAP_VALID |
+				SNDRV_PCM_INFO_BATCH,
 	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =		SNDRV_PCM_RATE_8000_48000,
 	.rate_min =		8000,
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index a299340519df..ce3f2e90f4d7 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -349,7 +349,8 @@ static struct snd_pcm_hardware snd_bt87x_digital_hw = {
 	.info = SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP_VALID,
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	.rates = 0, /* set at runtime */
 	.channels_min = 2,
@@ -365,7 +366,8 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = {
 	.info = SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP_VALID,
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
 	.rates = SNDRV_PCM_RATE_KNOT,
 	.rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX,
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index c7899c32aba1..449fe02f666e 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3014,7 +3014,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
 		.dev_free =	snd_cmipci_dev_free,
 	};
 	unsigned int val;
-	long iomidi;
+	long iomidi = 0;
 	int integrated_midi = 0;
 	char modelstr[16];
 	int pcm_index, pcm_spdif_index;
diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
index 3482ef69f491..2e44316530a2 100644
--- a/sound/pci/echoaudio/indigodjx.c
+++ b/sound/pci/echoaudio/indigodjx.c
@@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
 	.rates = 	SNDRV_PCM_RATE_32000 |
 			SNDRV_PCM_RATE_44100 |
 			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_64000 |
 			SNDRV_PCM_RATE_88200 |
 			SNDRV_PCM_RATE_96000,
 	.rate_min = 32000,
diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c
index aebee27a40ff..eb3819f9654a 100644
--- a/sound/pci/echoaudio/indigoiox.c
+++ b/sound/pci/echoaudio/indigoiox.c
@@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
 	.rates = 	SNDRV_PCM_RATE_32000 |
 			SNDRV_PCM_RATE_44100 |
 			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_64000 |
 			SNDRV_PCM_RATE_88200 |
 			SNDRV_PCM_RATE_96000,
 	.rate_min = 32000,
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 9bcd8ab5a27f..84cc49ca9148 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3817,6 +3817,49 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb ad1884a_mobile_verbs[] = {
+	/* DACs; unmute as default */
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+	/* Port-A (HP) mixer - route only from analog mixer */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* Port-A pin */
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Port-A (HP) pin - always unmuted */
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Port-B (mic jack) pin */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+	/* Port-C (int mic) pin */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+	/* Port-F (int speaker) mixer - route only from analog mixer */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* Port-F pin */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Analog mixer; mute as default */
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+	/* Analog Mix output amp */
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* capture sources */
+	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* unsolicited event for pin-sense */
+	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+	{ } /* end */
+};
+
 /*
  * Thinkpad X300
  * 0x11 - HP
@@ -3988,7 +4031,7 @@ static int patch_ad1884a(struct hda_codec *codec)
 		break;
 	case AD1884A_MOBILE:
 		spec->mixers[0] = ad1884a_mobile_mixers;
-		spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
+		spec->init_verbs[0] = ad1884a_mobile_verbs;
 		spec->multiout.dig_out_nid = 0;
 		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
 		codec->patch_ops.init = ad1884a_hp_init;
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8b79969034be..7cc38a11e997 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1238,7 +1238,8 @@ static struct snd_pcm_hardware snd_korg1212_playback_info =
 {
 	.info =              (SNDRV_PCM_INFO_MMAP |
                               SNDRV_PCM_INFO_MMAP_VALID |
-                              SNDRV_PCM_INFO_INTERLEAVED),
+			      SNDRV_PCM_INFO_INTERLEAVED |
+			      SNDRV_PCM_INFO_BATCH),
 	.formats =	      SNDRV_PCM_FMTBIT_S16_LE,
         .rates =              (SNDRV_PCM_RATE_44100 |
                               SNDRV_PCM_RATE_48000),
@@ -1258,7 +1259,8 @@ static struct snd_pcm_hardware snd_korg1212_capture_info =
 {
         .info =              (SNDRV_PCM_INFO_MMAP |
                               SNDRV_PCM_INFO_MMAP_VALID |
-                              SNDRV_PCM_INFO_INTERLEAVED),
+			      SNDRV_PCM_INFO_INTERLEAVED |
+			      SNDRV_PCM_INFO_BATCH),
         .formats =	      SNDRV_PCM_FMTBIT_S16_LE,
         .rates =	      (SNDRV_PCM_RATE_44100 |
                               SNDRV_PCM_RATE_48000),
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 01066c95580e..d057e6489643 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -240,7 +240,8 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
 static struct snd_pcm_hardware pdacf_pcm_capture_hw = {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
-				 SNDRV_PCM_INFO_MMAP_VALID),
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_BATCH),
 	.formats =		SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
 				SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
 				SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a259148..594c6c5b7838 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
 	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-			    SNDRV_PCM_INFO_INTERLEAVED,
+			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
 	.formats	  = AU1XPSC_PCM_FMTS,
 	.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
 	.period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 030d2454725f..f2653803ede8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
 obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..df7c8c281d2f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		4, 3, 0, output_tvl),
 
 	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+		TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
 	/* Common capture gain controls */
 	SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed9..0275321ff8ab 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
 	 * required for LRC in master mode. The DACs or ADCs need a
 	 * valid audio path i.e. pin -> ADC or DAC -> pin before
 	 * the LRC will be enabled in master mode. */
-	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+	if (!master || cmd != SNDRV_PCM_TRIGGER_START)
 		return 0;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f160fc..9f6be3d31ac0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value & 0xff;
-	int reg2 = (kcontrol->private_value >> 24) & 0xff;
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
 	int ret;
 	u16 val;
 
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+				    xinvert, tlv_array)			\
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
 		SNDRV_CTL_ELEM_ACCESS_READWRITE,  \
 	.tlv.p = (tlv_array), \
 	.info = snd_soc_info_volsw_2r, \
 	.get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
-	.private_value = (reg_left) | ((shift) << 8)  |		\
-		((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		.max = xmax, .invert = xinvert} }
 
 static const struct snd_kcontrol_new wm8580_snd_controls[] = {
 SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
 	reg = wm8580_read(codec, WM8580_PLLA4 + offset);
 	reg &= ~0x3f;
 	reg |= pll_div.prescale | pll_div.postscale << 1 |
-		pll_div.freqmode << 4;
+		pll_div.freqmode << 3;
 
 	wm8580_write(codec, WM8580_PLLA4 + offset, reg);
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..1111c710118a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 
 static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
 		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
 	.rate_min = 8000,
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 689ffcd17e1f..ab680aac3fcb 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -636,5 +636,6 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 
 	return snd_soc_register_dai(dai);
 }
-
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP_VALID),
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BATCH),
 	.formats =	DMABRG_FMTS,
 	.rates =	DMABRG_RATES,
 	.rate_min =		8000,
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index af95ff1e126c..1d2e51b3f918 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -1975,7 +1975,8 @@ static struct snd_pcm_hardware snd_dbri_pcm_hw = {
 	.info		= SNDRV_PCM_INFO_MMAP |
 			  SNDRV_PCM_INFO_INTERLEAVED |
 			  SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			  SNDRV_PCM_INFO_MMAP_VALID,
+			  SNDRV_PCM_INFO_MMAP_VALID |
+			  SNDRV_PCM_INFO_BATCH,
 	.formats	= SNDRV_PCM_FMTBIT_MU_LAW |
 			  SNDRV_PCM_FMTBIT_A_LAW |
 			  SNDRV_PCM_FMTBIT_U8 |
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 3f45c0fe61ab..b13ce767ac72 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -195,11 +195,14 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
 
 	debug("%s(%p)\n", __func__, substream);
 	
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dev->period_out_count[index] = BYTES_PER_SAMPLE + 1;
 		dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
-	else
+	} else {
+		dev->period_in_count[index] = BYTES_PER_SAMPLE;
 		dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE;
-	
+	}
+
 	if (dev->streaming)
 		return 0;
 	
@@ -300,8 +303,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev,
 		if (!sub)
 			continue;
 
-		pb = frames_to_bytes(sub->runtime, 
-				     sub->runtime->period_size);
+		pb = snd_pcm_lib_period_bytes(sub);
 		cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 					&dev->period_out_count[stream] :
 					&dev->period_in_count[stream];
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 6d517705da0e..515de1cd2a3e 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 9a608fa85155..dd1ab6177840 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -870,7 +870,8 @@ static struct snd_pcm_hardware snd_usX2Y_2c =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-				 SNDRV_PCM_INFO_MMAP_VALID),
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_BATCH),
 	.formats =                 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE,
 	.rates =                   SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
 	.rate_min =                44100,