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:
  sound: do not set DEVNAME for OSS devices
  ALSA: hda - Add sanity check in PCM open callback
  ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback
  ALSA: hda - Avoid invalid formats and rates with shared SPDIF
  ALSA: hda - Improve ASUS eeePC 1000 mixer
  ALSA: hda - Add GPIO1 control at muting with HP laptops
  ALSA: usx2y - reparent sound device
  ALSA: snd_usb_caiaq: reparent sound device
  sound: virtuoso: fix Xonar D1/DX silence after resume
  ASoC: Only disable pxa2xx-i2s clocks if we enabled them
  ALSA: hda - Add quirk for HP 6930p
  ALSA: hda - Add missing static to patch_ca0110()
  ASoC: OMAP: fix OMAP1510 broken PCM pointer callback
  ASoC: remove BROKEN from Efika and pcm030 fabric drivers
  ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig

13 files changed, 80 insertions, 40 deletions

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 462e2cedaa6a..26d255de6beb 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3470,10 +3470,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec,
                 }
                 mutex_lock(&codec->spdif_mutex);
                 if (mout->share_spdif) {
-                        runtime->hw.rates &= mout->spdif_rates;
-                        runtime->hw.formats &= mout->spdif_formats;
-                        if (mout->spdif_maxbps < hinfo->maxbps)
-                                hinfo->maxbps = mout->spdif_maxbps;
+                        if ((runtime->hw.rates & mout->spdif_rates) &&
+                            (runtime->hw.formats & mout->spdif_formats)) {
+                                runtime->hw.rates &= mout->spdif_rates;
+                                runtime->hw.formats &= mout->spdif_formats;
+                                if (mout->spdif_maxbps < hinfo->maxbps)
+                                        hinfo->maxbps = mout->spdif_maxbps;
+                        } else {
+                                mout->share_spdif = 0;
+                                /* FIXME: need notify? */
+                        }
                 }
                 mutex_unlock(&codec->spdif_mutex);
         }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4e9ea7080270..1877d95d4aa6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1454,6 +1454,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
                 mutex_unlock(&chip->open_mutex);
                 return err;
         }
+        snd_pcm_limit_hw_rates(runtime);
         spin_lock_irqsave(&chip->reg_lock, flags);
         azx_dev->substream = substream;
         azx_dev->running = 0;
@@ -1463,6 +1464,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
         snd_pcm_set_sync(substream);
         mutex_unlock(&chip->open_mutex);
 
+        if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max))
+                return -EINVAL;
+        if (snd_BUG_ON(!runtime->hw.formats))
+                return -EINVAL;
+        if (snd_BUG_ON(!runtime->hw.rates))
+                return -EINVAL;
         return 0;
 }
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1988582d1ab8..be7d25fa7f35 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3746,9 +3746,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
         { } /* end */
 };
 
+static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
+                                        struct snd_ctl_elem_value *ucontrol)
+{
+        struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+        int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+        int mute = (!ucontrol->value.integer.value[0] &&
+                    !ucontrol->value.integer.value[1]);
+        /* toggle GPIO1 according to the mute state */
+        snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+                            mute ? 0x02 : 0x0);
+        return ret;
+}
+
 static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
         HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-        HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+        /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "Master Playback Switch",
+                .info = snd_hda_mixer_amp_switch_info,
+                .get = snd_hda_mixer_amp_switch_get,
+                .put = ad1884a_mobile_master_sw_put,
+                .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+        },
         HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
         HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
         HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
@@ -3869,6 +3890,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = {
         /* 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},
+        /* allow to touch GPIO1 (for mute control) */
+        {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+        {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+        {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
         { } /* end */
 };
 
@@ -3978,6 +4003,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
         SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
         SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
         SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
+        SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
         SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
         SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
         SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 392d108c3558..019ca7cb56d7 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec)
 }
 
 
-int patch_ca0110(struct hda_codec *codec)
+static int patch_ca0110(struct hda_codec *codec)
 {
         struct ca0110_spec *spec;
         int err;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a8e58c483df..e661b21354be 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12876,20 +12876,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
         { }
 };
 
-/* bind volumes of both NID 0x0c and 0x0d */
-static struct hda_bind_ctls alc269_epc_bind_vol = {
-        .ops = &snd_hda_bind_vol,
-        .values = {
-                HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
-                HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
-                0
-        },
-};
-
 static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
-        HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-        HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
-        HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
         { } /* end */
 };
 
@@ -12902,12 +12893,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
 };
 
 /* FSC amilo */
-static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
-        HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-        HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-        HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol),
-        { } /* end */
-};
+#define alc269_fujitsu_mixer    alc269_eeepc_mixer
 
 static struct hda_verb alc269_quanta_fl1_verbs[] = {
         {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index bf971f7cfdc6..6ebcb6bdd712 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip)
 
 static void xonar_d1_resume(struct oxygen *chip)
 {
+        oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
+        msleep(1);
         cs43xx_init(chip);
         xonar_enable_output(chip);
 }
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 5dbebf82249c..8cb65ccad35f 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S
 config SND_SOC_MPC5200_AC97
         tristate "Freescale MPC5200 PSC in AC97 mode driver"
         depends on PPC_MPC52xx && PPC_BESTCOMM
-        select AC97_BUS
+        select SND_SOC_AC97_BUS
         select SND_MPC52xx_DMA
         select PPC_BESTCOMM_GEN_BD
         help
@@ -41,7 +41,7 @@ config SND_SOC_MPC5200_AC97
 
 config SND_MPC52xx_SOC_PCM030
         tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
-        depends on PPC_MPC5200_SIMPLE && BROKEN
+        depends on PPC_MPC5200_SIMPLE
         select SND_SOC_MPC5200_AC97
         select SND_SOC_WM9712
         help
@@ -50,7 +50,7 @@ config SND_MPC52xx_SOC_PCM030
 
 config SND_MPC52xx_SOC_EFIKA
         tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
-        depends on PPC_EFIKA && BROKEN
+        depends on PPC_EFIKA
         select SND_SOC_MPC5200_AC97
         select SND_SOC_STAC9766
         help
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 6454e15f7d28..84a1950880eb 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
         dma_addr_t ptr;
         snd_pcm_uframes_t offset;
 
-        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-                ptr = omap_get_dma_src_pos(prtd->dma_ch);
-        else
+        if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
                 ptr = omap_get_dma_dst_pos(prtd->dma_ch);
+                offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+        } else if (!(cpu_is_omap1510())) {
+                ptr = omap_get_dma_src_pos(prtd->dma_ch);
+                offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+        } else
+                offset = prtd->period_index * runtime->period_size;
 
-        offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
         if (offset >= runtime->buffer_size)
                 offset = 0;
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 4743e262895d..6b8f655d1ad8 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -167,6 +167,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
         BUG_ON(IS_ERR(clk_i2s));
         clk_enable(clk_i2s);
+        dai->private_data = dai;
         pxa_i2s_wait();
 
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -255,7 +256,10 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
         if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
                 SACR0 &= ~SACR0_ENB;
                 pxa_i2s_wait();
-                clk_disable(clk_i2s);
+                if (dai->private_data != NULL) {
+                        clk_disable(clk_i2s);
+                        dai->private_data = NULL;
+                }
         }
 }
 
@@ -336,6 +340,7 @@ static int pxa2xx_i2s_probe(struct platform_device *dev)
                 return PTR_ERR(clk_i2s);
 
         pxa_i2s_dai.dev = &dev->dev;
+        pxa_i2s_dai.private_data = NULL;
         ret = snd_soc_register_dai(&pxa_i2s_dai);
         if (ret != 0)
                 clk_put(clk_i2s);
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 12522e6913d9..a41f8b127f49 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -10,6 +10,8 @@
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/err.h>
+#include <linux/kdev_t.h>
+#include <linux/major.h>
 #include <sound/core.h>
 
 #ifdef CONFIG_SOUND_OSS_CORE
@@ -29,6 +31,8 @@ MODULE_LICENSE("GPL");
 
 static char *sound_nodename(struct device *dev)
 {
+        if (MAJOR(dev->devt) == SOUND_MAJOR)
+                return NULL;
         return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev));
 }
 
@@ -104,7 +108,6 @@ module_exit(cleanup_soundcore);
 #include <linux/types.h>
 #include <linux/kernel.h>
 #include <linux/sound.h>
-#include <linux/major.h>
 #include <linux/kmod.h>
 
 #define SOUND_STEP 16
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 0e5db719de24..de38108f0b28 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 1.3.17");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
                          "{Native Instruments, RigKontrol3},"
@@ -349,7 +349,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
                 log("Unable to set up control system (ret=%d)\n", ret);
 }
 
-static int create_card(struct usb_device* usb_dev, struct snd_card **cardp)
+static int create_card(struct usb_device *usb_dev,
+                       struct usb_interface *intf,
+                       struct snd_card **cardp)
 {
         int devnum;
         int err;
@@ -374,7 +376,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp)
         dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor),
                                   le16_to_cpu(usb_dev->descriptor.idProduct));
         spin_lock_init(&dev->spinlock);
-        snd_card_set_dev(card, &usb_dev->dev);
+        snd_card_set_dev(card, &intf->dev);
 
         *cardp = card;
         return 0;
@@ -461,7 +463,7 @@ static int __devinit snd_probe(struct usb_interface *intf,
         struct snd_card *card;
         struct usb_device *device = interface_to_usbdev(intf);
 
-        ret = create_card(device, &card);
+        ret = create_card(device, intf, &card);
 
         if (ret < 0)
                 return ret;
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index a5aae9d67f31..fd44946ce4b3 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -514,7 +514,6 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp)
                 US122L(card)->chip.dev->bus->busnum,
                 US122L(card)->chip.dev->devnum
                 );
-        snd_card_set_dev(card, &device->dev);
         *cardp = card;
         return 0;
 }
@@ -531,6 +530,7 @@ static int us122l_usb_probe(struct usb_interface *intf,
         if (err < 0)
                 return err;
 
+        snd_card_set_dev(card, &intf->dev);
         if (!us122l_create_card(card)) {
                 snd_card_free(card);
                 return -EINVAL;
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 5ce0da23ee96..cb4bb8373ca2 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -364,7 +364,6 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp)
                 0,//us428(card)->usbmidi.ifnum,
                 usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum
                 );
-        snd_card_set_dev(card, &device->dev);
         *cardp = card;
         return 0;
 }
@@ -388,6 +387,7 @@ static int usX2Y_usb_probe(struct usb_device *device,
         err = usX2Y_create_card(device, &card);
         if (err < 0)
                 return err;
+        snd_card_set_dev(card, &intf->dev);
         if ((err = usX2Y_hwdep_new(card, device)) < 0  ||
             (err = snd_card_register(card)) < 0) {
                 snd_card_free(card);