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: hda - Set function_id only on FG nodes
  ALSA: MAINTAINERS - Update SOUND
  ALSA: emu10k1 - off by 1 in snd_emu10k1_wait()
  ASoC: OMAP: Fix FS polarity in OSK5912 machine driver
  ASoC: OMAP: Fix DSP_B format in OMAP McBSP DAI driver
  ASoC: Fix include build error in s3c2412-i2s.c
  ASoC: Fix s3c-i2s-v2.c snd_soc_dai changes
  ASoC: s3c-i2s-v2.c fix for s3c_i2sv2_iis_calc_rate
  ASoC: Fix jive_wm8750.c build problems
  ASoC: pxa-ssp: allow setting of dai format 0
  ALSA: hda - Add upper-limit of mixer amp for AD1884A-laptop model, too
  ALSA: hda - Fix headphone-detection on some machines with STAC/IDT codecs
  ALSA: Intel8x0: Add hp_only quirk for SSID 0x1028016a (Dell Inspiron 8600)
  ALSA: Intel8x0: Remove conflicting quirk for SSID 0x103c0934
  ALSA: hda_intel.c - Consolidate bitfields

MAINTAINERS

@@ -5235,7 +5235,12 @@ M:	perex@perex.cz
 P:	Takashi Iwai
 M:	tiwai@suse.de
 L:	alsa-devel@alsa-project.org (subscribers-only)
+W:	http://www.alsa-project.org/
+T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
+T:	git git://git.alsa-project.org/alsa-kernel.git
 S:	Maintained
+F:	Documentation/sound/
+F:	include/sound/
 F:	sound/
 
 SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)

sound/pci/emu10k1/io.c

@@ -490,7 +490,7 @@ void snd_emu10k1_wait(struct snd_emu10k1 *emu, unsigned int wait)
 			if (newtime != curtime)
 				break;
 		}
-		if (count >= 16384)
+		if (count > 16384)
 			break;
 		curtime = newtime;
 	}

sound/pci/hda/hda_codec.c

@@ -642,19 +642,21 @@ static int get_codec_name(struct hda_codec *codec)
  */
 static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec)
 {
-	int i, total_nodes;
+	int i, total_nodes, function_id;
 	hda_nid_t nid;
 
 	total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid);
 	for (i = 0; i < total_nodes; i++, nid++) {
-		codec->function_id = snd_hda_param_read(codec, nid,
+		function_id = snd_hda_param_read(codec, nid,
 						AC_PAR_FUNCTION_TYPE) & 0xff;
-		switch (codec->function_id) {
+		switch (function_id) {
 		case AC_GRP_AUDIO_FUNCTION:
 			codec->afg = nid;
+			codec->function_id = function_id;
 			break;
 		case AC_GRP_MODEM_FUNCTION:
 			codec->mfg = nid;
+			codec->function_id = function_id;
 			break;
 		default:
 			break;

sound/pci/hda/hda_intel.c

@@ -312,7 +312,6 @@ struct azx_dev {
 	unsigned int period_bytes; /* size of the period in bytes */
 	unsigned int frags;	/* number for period in the play buffer */
 	unsigned int fifo_size;	/* FIFO size */
-	unsigned int start_flag: 1;	/* stream full start flag */
 	unsigned long start_jiffies;	/* start + minimum jiffies */
 	unsigned long min_jiffies;	/* minimum jiffies before position is valid */
 
@@ -333,6 +332,7 @@ struct azx_dev {
 	unsigned int opened :1;
 	unsigned int running :1;
 	unsigned int irq_pending :1;
+	unsigned int start_flag: 1;	/* stream full start flag */
 	/*
 	 * For VIA:
 	 *  A flag to ensure DMA position is 0

sound/pci/hda/patch_analog.c

@@ -3977,6 +3977,14 @@ static int patch_ad1884a(struct hda_codec *codec)
 		spec->input_mux = &ad1884a_laptop_capture_source;
 		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
 		codec->patch_ops.init = ad1884a_hp_init;
+		/* set the upper-limit for mixer amp to 0dB for avoiding the
+		 * possible damage by overloading
+		 */
+		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
+					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+					  (1 << AC_AMPCAP_MUTE_SHIFT));
 		break;
 	case AD1884A_MOBILE:
 		spec->mixers[0] = ad1884a_mobile_mixers;

sound/pci/hda/patch_sigmatel.c

@@ -3076,6 +3076,11 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 	unsigned int wid_caps;
 
 	for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) {
+		if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) {
+			wid_caps = get_wcaps(codec, pins[i]);
+			if (wid_caps & AC_WCAP_UNSOL_CAP)
+				spec->hp_detect = 1;
+		}
 		nid = dac_nids[i];
 		if (!nid)
 			continue;
@@ -3119,11 +3124,6 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 			err = create_controls_idx(codec, name, idx, nid, 3);
 			if (err < 0)
 				return err;
-			if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) {
-				wid_caps = get_wcaps(codec, pins[i]);
-				if (wid_caps & AC_WCAP_UNSOL_CAP)
-					spec->hp_detect = 1;
-			}
 		}
 	}
 	return 0;

sound/pci/intel8x0.c

@@ -1852,6 +1852,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 		.name = "Dell Unknown",	/* STAC9750/51 */
 		.type = AC97_TUNE_HP_ONLY
 	},
+	{
+		.subvendor = 0x1028,
+		.subdevice = 0x016a,
+		.name = "Dell Inspiron 8600",	/* STAC9750/51 */
+		.type = AC97_TUNE_HP_ONLY
+	},
 	{
 		.subvendor = 0x1028,
 		.subdevice = 0x0186,
@@ -1894,12 +1900,6 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 		.name = "HP nc6000",
 		.type = AC97_TUNE_MUTE_LED
 	},
-	{
-		.subvendor = 0x103c,
-		.subdevice = 0x0934,
-		.name = "HP nx8220",
-		.type = AC97_TUNE_MUTE_LED
-	},
 	{
 		.subvendor = 0x103c,
 		.subdevice = 0x129d,

sound/soc/omap/omap-mcbsp.c

@@ -283,7 +283,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		regs->srgr2	|= FPER(wlen * channels - 1);
-		regs->srgr1	|= FWID(wlen * channels - 2);
+		regs->srgr1	|= FWID(0);
 		break;
 	}
 
@@ -302,6 +302,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 {
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+	unsigned int temp_fmt = fmt;
 
 	if (mcbsp_data->configured)
 		return 0;
@@ -328,6 +329,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* 0-bit data delay */
 		regs->rcr2      |= RDATDLY(0);
 		regs->xcr2      |= XDATDLY(0);
+		/* Invert FS polarity configuration */
+		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
 		break;
 	default:
 		/* Unsupported data format */
@@ -351,7 +354,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	}
 
 	/* Set bit clock (CLKX/CLKR) and FS polarities */
-	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
 		/*
 		 * Normal BCLK + FS.

sound/soc/omap/osk5912.c

@@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 	/* Set codec DAI configuration */
 	err = snd_soc_dai_set_fmt(codec_dai,
 				  SND_SOC_DAIFMT_DSP_B |
-				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_NB_NF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
 		printk(KERN_ERR "can't set codec DAI configuration\n");
@@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 	/* Set cpu DAI configuration */
 	err = snd_soc_dai_set_fmt(cpu_dai,
 				  SND_SOC_DAIFMT_DSP_B |
-				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_NB_NF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
 		printk(KERN_ERR "can't set cpu DAI configuration\n");

sound/soc/pxa/pxa-ssp.c

@@ -806,6 +806,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
 		goto err_priv;
 	}
 
+	priv->dai_fmt = (unsigned int) -1;
 	dai->private_data = priv;
 
 	return 0;

sound/soc/s3c24xx/jive_wm8750.c

@@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
-			    s3c2412_get_iisclk());
+	s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
+				s3c2412_get_iisclk());
 
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
@@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = {
 };
 
 /* jive audio machine driver */
-static struct snd_soc_machine snd_soc_machine_jive = {
+static struct snd_soc_card snd_soc_machine_jive = {
 	.name		= "Jive",
+	.platform	= &s3c24xx_soc_platform,
 	.dai_link	= &jive_dai,
 	.num_links	= 1,
 };
@@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = {
 
 /* jive audio subsystem */
 static struct snd_soc_device jive_snd_devdata = {
-	.machine	= &snd_soc_machine_jive,
-	.platform	= &s3c24xx_soc_platform,
-	.codec_dev	= &soc_codec_dev_wm8750_spi,
+	.card		= &snd_soc_machine_jive,
+	.codec_dev	= &soc_codec_dev_wm8750,
 	.codec_data	= &jive_wm8750_setup,
 };
 

sound/soc/s3c24xx/s3c-i2s-v2.c

@@ -473,9 +473,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 /* default table of all avaialable root fs divisors */
 static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
 
-int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
-			  unsigned int *fstab,
-			  unsigned int rate, struct clk *clk)
+int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+			    unsigned int *fstab,
+			    unsigned int rate, struct clk *clk)
 {
 	unsigned long clkrate = clk_get_rate(clk);
 	unsigned int div;
@@ -531,7 +531,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
 
 	return 0;
 }
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate);
 
 int s3c_i2sv2_probe(struct platform_device *pdev,
 		    struct snd_soc_dai *dai,
@@ -624,10 +624,12 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 
 int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 {
-	dai->ops.trigger = s3c2412_i2s_trigger;
-	dai->ops.hw_params = s3c2412_i2s_hw_params;
-	dai->ops.set_fmt = s3c2412_i2s_set_fmt;
-	dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+	struct snd_soc_dai_ops *ops = dai->ops;
+
+	ops->trigger = s3c2412_i2s_trigger;
+	ops->hw_params = s3c2412_i2s_hw_params;
+	ops->set_fmt = s3c2412_i2s_set_fmt;
+	ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
 
 	dai->suspend = s3c2412_i2s_suspend;
 	dai->resume = s3c2412_i2s_resume;

sound/soc/s3c24xx/s3c2412-i2s.c

@@ -33,8 +33,8 @@
 
 #include <plat/regs-s3c2412-iis.h>
 
-#include <plat/regs-gpio.h>
 #include <plat/audio.h>
+#include <mach/regs-gpio.h>
 #include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"