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] soc - wm9712 - checkpatch fixes
  [ALSA] pcsp - Fix more dependency
  [ALSA] hda - Add support of Medion RIM 2150
  [ALSA] ASoC: Add drivers for the Texas Instruments OMAP processors
  [ALSA] ice1724 - Enable watermarks
  [ALSA] Add MPU401_INFO_NO_ACK bitflag

Documentation/sound/alsa/ALSA-Configuration.txt

@@ -795,6 +795,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  lg-lw		LG LW20/LW25 laptop
 	  tcl		TCL S700
 	  clevo		Clevo laptops (m520G, m665n)
+	  medion	Medion Rim 2150
 	  test		for testing/debugging purpose, almost all controls can be
 			adjusted.  Appearing only when compiled with
 			$CONFIG_SND_DEBUG=y

include/sound/mpu401.h

@@ -50,6 +50,7 @@
 #define MPU401_INFO_INTEGRATED	(1 << 2)	/* integrated h/w port */
 #define MPU401_INFO_MMIO	(1 << 3)	/* MMIO access */
 #define MPU401_INFO_TX_IRQ	(1 << 4)	/* independent TX irq */
+#define MPU401_INFO_NO_ACK	(1 << 6)	/* No ACK cmd needed */
 
 #define MPU401_MODE_BIT_INPUT		0
 #define MPU401_MODE_BIT_OUTPUT		1

sound/drivers/Kconfig

@@ -8,6 +8,8 @@ config SND_PCSP
 	tristate "Internal PC speaker support"
 	depends on X86_PC && HIGH_RES_TIMERS
 	depends on INPUT
+	depends on SND
+	select SND_PCM
 	help
 	  If you don't have a sound card in your computer, you can include a
 	  driver for the PC speaker which allows it to act like a primitive

sound/drivers/mpu401/mpu401_uart.c

@@ -243,7 +243,7 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd,
 #endif
 	}
 	mpu->write(mpu, cmd, MPU401C(mpu));
-	if (ack) {
+	if (ack && !(mpu->info_flags & MPU401_INFO_NO_ACK)) {
 		ok = 0;
 		timeout = 10000;
 		while (!ok && timeout-- > 0) {

sound/pci/hda/patch_realtek.c

@@ -60,6 +60,7 @@ enum {
 	ALC880_TCL_S700,
 	ALC880_LG,
 	ALC880_LG_LW,
+	ALC880_MEDION_RIM,
 #ifdef CONFIG_SND_DEBUG
 	ALC880_TEST,
 #endif
@@ -2275,6 +2276,75 @@ static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
 		alc880_lg_lw_automute(codec);
 }
 
+static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct hda_input_mux alc880_medion_rim_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+	},
+};
+
+static struct hda_verb alc880_medion_rim_init_verbs[] = {
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Mic2 (as headphone out) for HP output */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Internal Speaker */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x3060},
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_medion_rim_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	if (present)
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
+	else
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
+}
+
+static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
+					  unsigned int res)
+{
+	/* Looks like the unsol event is incompatible with the standard
+	 * definition.  4bit tag is placed at 28 bit!
+	 */
+	if ((res >> 28) == ALC880_HP_EVENT)
+		alc880_medion_rim_automute(codec);
+}
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static struct hda_amp_list alc880_loopbacks[] = {
 	{ 0x0b, HDA_INPUT, 0 },
@@ -2882,6 +2952,7 @@ static const char *alc880_models[ALC880_MODEL_LAST] = {
 	[ALC880_F1734]		= "F1734",
 	[ALC880_LG]		= "lg",
 	[ALC880_LG_LW]		= "lg-lw",
+	[ALC880_MEDION_RIM]	= "medion",
 #ifdef CONFIG_SND_DEBUG
 	[ALC880_TEST]		= "test",
 #endif
@@ -2933,6 +3004,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
 	SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
 	SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
+	SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
 	SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
@@ -3227,6 +3299,20 @@ static struct alc_config_preset alc880_presets[] = {
 		.unsol_event = alc880_lg_lw_unsol_event,
 		.init_hook = alc880_lg_lw_automute,
 	},
+	[ALC880_MEDION_RIM] = {
+		.mixers = { alc880_medion_rim_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_medion_rim_init_verbs,
+				alc_gpio2_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_medion_rim_capture_source,
+		.unsol_event = alc880_medion_rim_unsol_event,
+		.init_hook = alc880_medion_rim_automute,
+	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC880_TEST] = {
 		.mixers = { alc880_test_mixer },

sound/pci/ice1712/ice1724.c

@@ -2429,6 +2429,7 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 			if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
 						       ICEREG1724(ice, MPU_CTRL),
 						       (MPU401_INFO_INTEGRATED |
+							MPU401_INFO_NO_ACK |
 							MPU401_INFO_TX_IRQ),
 						       ice->irq, 0,
 						       &ice->rmidi[0])) < 0) {
@@ -2442,12 +2443,10 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 			outb(inb(ICEREG1724(ice, IRQMASK)) &
 			     ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX),
 			     ICEREG1724(ice, IRQMASK));
-#if 0 /* for testing */
 			/* set watermarks */
 			outb(VT1724_MPU_RX_FIFO | 0x1,
 			     ICEREG1724(ice, MPU_FIFO_WM));
 			outb(0x1, ICEREG1724(ice, MPU_FIFO_WM));
-#endif
 		}
 	}
 

sound/soc/Kconfig

@@ -30,6 +30,7 @@ source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/sh/Kconfig"
 source "sound/soc/fsl/Kconfig"
 source "sound/soc/davinci/Kconfig"
+source "sound/soc/omap/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"

sound/soc/Makefile

@@ -1,4 +1,4 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
-obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
+obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/

sound/soc/codecs/wm9712.c

@@ -37,23 +37,23 @@ static int ac97_write(struct snd_soc_codec *codec,
  * WM9712 register cache
  */
 static const u16 wm9712_reg[] = {
-	0x6174, 0x8000, 0x8000, 0x8000, // 6
-	0x0f0f, 0xaaa0, 0xc008, 0x6808, // e
-	0xe808, 0xaaa0, 0xad00, 0x8000, // 16
-	0xe808, 0x3000, 0x8000, 0x0000, // 1e
-	0x0000, 0x0000, 0x0000, 0x000f, // 26
-	0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
-	0x0000, 0xbb80, 0x0000, 0x0000, // 36
-	0x0000, 0x2000, 0x0000, 0x0000, // 3e
-	0x0000, 0x0000, 0x0000, 0x0000, // 46
-	0x0000, 0x0000, 0xf83e, 0xffff, // 4e
-	0x0000, 0x0000, 0x0000, 0xf83e, // 56
-	0x0008, 0x0000, 0x0000, 0x0000, // 5e
-	0xb032, 0x3e00, 0x0000, 0x0000, // 66
-	0x0000, 0x0000, 0x0000, 0x0000, // 6e
-	0x0000, 0x0000, 0x0000, 0x0006, // 76
-	0x0001, 0x0000, 0x574d, 0x4c12, // 7e
-	0x0000, 0x0000 // virtual hp mixers
+	0x6174, 0x8000, 0x8000, 0x8000, /*  6 */
+	0x0f0f, 0xaaa0, 0xc008, 0x6808, /*  e */
+	0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */
+	0xe808, 0x3000, 0x8000, 0x0000, /* 1e */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+	0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+	0x0000, 0x2000, 0x0000, 0x0000, /* 3e */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 46 */
+	0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */
+	0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */
+	0x0008, 0x0000, 0x0000, 0x0000, /* 5e */
+	0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+	0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+	0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */
+	0x0000, 0x0000 /* virtual hp mixers */
 };
 
 /* virtual HP mixers regs */
@@ -94,7 +94,7 @@ static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
 SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
 SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
 SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
-SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
 SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
 
 SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
@@ -165,7 +165,8 @@ static int wm9712_add_controls(struct snd_soc_codec *codec)
 
 	for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
 		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL));
+				  snd_soc_cnew(&wm9712_snd_ac97_controls[i],
+					       codec, NULL));
 		if (err < 0)
 			return err;
 	}
@@ -363,7 +364,6 @@ static const char *audio_map[][3] = {
 	{"Left HP Mixer", "PCM Playback Switch",  "Left DAC"},
 	{"Left HP Mixer", "Mic Sidetone Switch",  "Mic PGA"},
 	{"Left HP Mixer", NULL,  "ALC Sidetone Mux"},
-	//{"Right HP Mixer", NULL, "HP Mixer"},
 
 	/* Right HP mixer */
 	{"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
@@ -454,15 +454,13 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec)
 {
 	int i;
 
-	for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) {
+	for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++)
 		snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
-	}
 
-	/* set up audio path audio_mapnects */
-	for(i = 0; audio_map[i][0] != NULL; i++) {
+	/* set up audio path connects */
+	for (i = 0; audio_map[i][0] != NULL; i++)
 		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+					   audio_map[i][1], audio_map[i][2]);
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -540,7 +538,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 }
 
 #define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
+		SNDRV_PCM_RATE_48000)
 
 struct snd_soc_codec_dai wm9712_dai[] = {
 {
@@ -577,8 +576,6 @@ EXPORT_SYMBOL_GPL(wm9712_dai);
 
 static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
 {
-	u16 reg;
-
 	switch (event) {
 	case SNDRV_CTL_POWER_D0: /* full On */
 	case SNDRV_CTL_POWER_D1: /* partial On */
@@ -633,7 +630,7 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 	u16 *cache = codec->reg_cache;
 
 	ret = wm9712_reset(codec, 1);
-	if (ret < 0){
+	if (ret < 0) {
 		printk(KERN_ERR "could not reset AC97 codec\n");
 		return ret;
 	}
@@ -642,9 +639,9 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 
 	if (ret == 0) {
 		/* Sync reg_cache with the hardware after cold reset */
-		for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) {
+		for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) {
 			if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
-				(i > 0x58 && i != 0x5c))
+			    (i > 0x58 && i != 0x5c))
 				continue;
 			soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
 		}
@@ -757,7 +754,6 @@ struct snd_soc_codec_device soc_codec_dev_wm9712 = {
 	.suspend =	wm9712_soc_suspend,
 	.resume =	wm9712_soc_resume,
 };
-
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
 
 MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");

sound/soc/omap/Kconfig

@@ -0,0 +1,19 @@
+menu "SoC Audio for the Texas Instruments OMAP"
+
+config SND_OMAP_SOC
+	tristate "SoC Audio for the Texas Instruments OMAP chips"
+	depends on ARCH_OMAP && SND_SOC
+
+config SND_OMAP_SOC_MCBSP
+	tristate
+	select OMAP_MCBSP
+
+config SND_OMAP_SOC_N810
+	tristate "SoC Audio support for Nokia N810"
+	depends on SND_OMAP_SOC && MACH_NOKIA_N810
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on Nokia N810.
+
+endmenu

sound/soc/omap/Makefile

@@ -0,0 +1,11 @@
+# OMAP Platform Support
+snd-soc-omap-objs := omap-pcm.o
+snd-soc-omap-mcbsp-objs := omap-mcbsp.o
+
+obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
+obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+
+# OMAP Machine Support
+snd-soc-n810-objs := n810.o
+
+obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o