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: Fix Trivial Warnining in sound/pci/cmipci.c
  ALSA: snd-usb-caiaq: fix reported elapsed periods
  ASoC: s3c-i2s-v2 needs to declare a license for modular builds
  ALSA: hda - Fix init verbs of AD1884A mobile model
  ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991
  ASoC: Fix WM8580 volume update handling for large register changes
  ASoC: Fix offset of freqmode in WM8580 PLL configuration

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;

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;

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

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);
 

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");

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];

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},"