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

13 files changed, 56 insertions, 35 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index 0beac8a7f8f2..1e067a675e53 100644
--- a/MAINTAINERS
+++ b/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)
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index 4bfc31d1b281..c1a5aa15af8f 100644
--- a/sound/pci/emu10k1/io.c
+++ b/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;
         }
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index fd6e6f337d10..8820faf6c9d8 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/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;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index bc882f8f163c..21e99cfa8c49 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/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
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 38ad3f7b040f..9bcd8ab5a27f 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/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;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index ce30b459aee6..917bc5d3ac2c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/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;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 5dced5b79387..8042d5398892 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1854,6 +1854,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
         },
         {
                 .subvendor = 0x1028,
+                .subdevice = 0x016a,
+                .name = "Dell Inspiron 8600",   /* STAC9750/51 */
+                .type = AC97_TUNE_HP_ONLY
+        },
+        {
+                .subvendor = 0x1028,
                 .subdevice = 0x0186,
                 .name = "Dell Latitude D810", /* cf. Malone #41015 */
                 .type = AC97_TUNE_HP_MUTE_LED
@@ -1896,12 +1902,6 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
         },
         {
                 .subvendor = 0x103c,
-                .subdevice = 0x0934,
-                .name = "HP nx8220",
-                .type = AC97_TUNE_MUTE_LED
-        },
-        {
-                .subvendor = 0x103c,
                 .subdevice = 0x129d,
                 .name = "HP xw8000",
                 .type = AC97_TUNE_HP_ONLY
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 9c09b94f0cf8..90f4df7fd906 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/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.
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index a952a4eb3361..a4e149b7f0eb 100644
--- a/sound/soc/omap/osk5912.c
+++ b/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");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 308a657928d2..de2254475d52 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/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;
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 32063790d95b..93e6c87b7399 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/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,
 };
 
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 295a4c910262..689ffcd17e1f 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/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;
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 1ca3cdaa8213..b7e0b3f0bfc8 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/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"