Merge branch 'akpm' (patches from Andrew)

Merge yet more updates from Andrew Morton:

 - a few MM remainders

 - misc things

 - autofs updates

 - signals

 - affs updates

 - ipc

 - nilfs2

 - spelling.txt updates

* emailed patches from Andrew Morton <akpm@linux-foundation.org>: (78 commits)
  mm, x86: fix HIGHMEM64 && PARAVIRT build config for native_pud_clear()
  mm: add arch-independent testcases for RODATA
  hfs: atomically read inode size
  mm: clarify mm_struct.mm_{users,count} documentation
  mm: use mmget_not_zero() helper
  mm: add new mmget() helper
  mm: add new mmgrab() helper
  checkpatch: warn when formats use %Z and suggest %z
  lib/vsprintf.c: remove %Z support
  scripts/spelling.txt: add some typo-words
  scripts/spelling.txt: add "followings" pattern and fix typo instances
  scripts/spelling.txt: add "therfore" pattern and fix typo instances
  scripts/spelling.txt: add "overwriten" pattern and fix typo instances
  scripts/spelling.txt: add "overwritting" pattern and fix typo instances
  scripts/spelling.txt: add "deintialize(d)" pattern and fix typo instances
  scripts/spelling.txt: add "disassocation" pattern and fix typo instances
  scripts/spelling.txt: add "omited" pattern and fix typo instances
  scripts/spelling.txt: add "explictely" pattern and fix typo instances
  scripts/spelling.txt: add "applys" pattern and fix typo instances
  scripts/spelling.txt: add "configuartion" pattern and fix typo instances
  ...

15 files changed, 17 insertions, 17 deletions

diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index f4234ed..8cf0dc7 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -3093,7 +3093,7 @@ static int patch_cm9739(struct snd_ac97 * ac97)
 	/* set-up multi channel */
 	/* bit 14: 0 = SPDIF, 1 = EAPD */
 	/* bit 13: enable internal vref output for mic */
-	/* bit 12: disable center/lfe (swithable) */
+	/* bit 12: disable center/lfe (switchable) */
 	/* bit 10: disable surround/line (switchable) */
 	/* bit 9: mix 2 surround off */
 	/* bit 4: undocumented; 0 mutes the CM9739A, which defaults to 1 */
diff --git a/sound/pci/cs46xx/cs46xx_dsp_task_types.h b/sound/pci/cs46xx/cs46xx_dsp_task_types.h
index 5cf920b..be56947 100644
--- a/sound/pci/cs46xx/cs46xx_dsp_task_types.h
+++ b/sound/pci/cs46xx/cs46xx_dsp_task_types.h
@@ -203,7 +203,7 @@ struct dsp_task_tree_context_block {
 
 	u32	  saverfe;					
 
-	/* Value may be overwriten by stack save algorithm.
+	/* Value may be overwritten by stack save algorithm.
 	   Retain the size of the stack data saved here if used */
 	___DSP_DUAL_16BIT_ALLOC(
              reserved1,	
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 9ec4dba..07a9deb 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2866,7 +2866,7 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
 #define CA0132_CODEC_MUTE(xname, nid, dir) \
 	CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir)
 
-/* The followings are for tuning of products */
+/* The following are for tuning of products */
 #ifdef ENABLE_TUNING_CONTROLS
 
 static unsigned int voice_focus_vals_lookup[] = {
diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c
index f7ac8d5..27c03e4 100644
--- a/sound/pci/ice1712/wm8766.c
+++ b/sound/pci/ice1712/wm8766.c
@@ -254,7 +254,7 @@ static int snd_wm8766_ctl_put(struct snd_kcontrol *kcontrol,
 	int n = kcontrol->private_value;
 	u16 val, regval1, regval2;
 
-	/* this also works for enum because value is an union */
+	/* this also works for enum because value is a union */
 	regval1 = ucontrol->value.integer.value[0];
 	regval2 = ucontrol->value.integer.value[1];
 	if (wm->ctl[n].flags & WM8766_FLAG_INVERT) {
diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c
index ebd2fe4..553669b 100644
--- a/sound/pci/ice1712/wm8776.c
+++ b/sound/pci/ice1712/wm8776.c
@@ -528,7 +528,7 @@ static int snd_wm8776_ctl_put(struct snd_kcontrol *kcontrol,
 	int n = kcontrol->private_value;
 	u16 val, regval1, regval2;
 
-	/* this also works for enum because value is an union */
+	/* this also works for enum because value is a union */
 	regval1 = ucontrol->value.integer.value[0];
 	regval2 = ucontrol->value.integer.value[1];
 	if (wm->ctl[n].flags & WM8776_FLAG_INVERT) {
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 565f7f5..1e25095 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2051,7 +2051,7 @@ static void snd_korg1212_proc_read(struct snd_info_entry *entry,
 	snd_iprintf(buffer, korg1212->card->longname);
 	snd_iprintf(buffer, " (index #%d)\n", korg1212->card->number + 1);
 	snd_iprintf(buffer, "\nGeneral settings\n");
-	snd_iprintf(buffer, "    period size: %Zd bytes\n", K1212_PERIOD_BYTES);
+	snd_iprintf(buffer, "    period size: %zd bytes\n", K1212_PERIOD_BYTES);
 	snd_iprintf(buffer, "     clock mode: %s\n", clockSourceName[korg1212->clkSrcRate] );
 	snd_iprintf(buffer, "  left ADC Sens: %d\n", korg1212->leftADCInSens );
 	snd_iprintf(buffer, " right ADC Sens: %d\n", korg1212->rightADCInSens );
@@ -2276,7 +2276,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci,
 
 	if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
 				sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0) {
-		snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%Zd bytes)\n", sizeof(struct KorgSharedBuffer));
+		snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%zd bytes)\n", sizeof(struct KorgSharedBuffer));
                 snd_korg1212_free(korg1212);
                 return -ENOMEM;
         }
diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c
index 8063305..a99808a 100644
--- a/sound/pci/pcxhr/pcxhr_hwdep.c
+++ b/sound/pci/pcxhr/pcxhr_hwdep.c
@@ -292,7 +292,7 @@ static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index,
 	int err, card_index;
 
 	dev_dbg(&mgr->pci->dev,
-		"loading dsp [%d] size = %Zd\n", index, dsp->size);
+		"loading dsp [%d] size = %zd\n", index, dsp->size);
 
 	switch (index) {
 	case PCXHR_FIRMWARE_XLX_INT_INDEX:
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 56aa1ba..5f97791 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -201,7 +201,7 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware *
 	c |= (int)vx_inb(chip, RXM) << 8;
 	c |= vx_inb(chip, RXL);
 
-	snd_printdd(KERN_DEBUG "xilinx: dsp size received 0x%x, orig 0x%Zx\n", c, fw->size);
+	snd_printdd(KERN_DEBUG "xilinx: dsp size received 0x%x, orig 0x%zx\n", c, fw->size);
 
 	vx_outb(chip, ICR, ICR_HF0);
 
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index b84d7d3..cdd44ab 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -883,7 +883,7 @@ static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start)
 static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card)
 {
 	/*
-	 * avsetting driver seems to never change the followings
+	 * avsetting driver seems to never change the following
 	 * so, init them here once
 	 */
 
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 818b052..ec1067a 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -506,7 +506,7 @@ static int acp_init(void __iomem *acp_mmio)
 	return 0;
 }
 
-/* Deintialize ACP */
+/* Deinitialize ACP */
 static int acp_deinit(void __iomem *acp_mmio)
 {
 	u32 val;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 624b3b9..63b2745 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1269,7 +1269,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_ON:
-		/* Turn off any unneded single ended outputs */
+		/* Turn off any unneeded single ended outputs */
 		val = 0;
 		mask = 0;
 
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 1d82f68..8cfffa7 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -368,7 +368,7 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
 	fsl_asrc_set_watermarks(pair, ASRC_INPUTFIFO_THRESHOLD,
 				ASRC_INPUTFIFO_THRESHOLD);
 
-	/* Configure the followings only for Ideal Ratio mode */
+	/* Configure the following only for Ideal Ratio mode */
 	if (!ideal)
 		return 0;
 
diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h
index 924971b..9b03135 100644
--- a/sound/soc/qcom/lpass.h
+++ b/sound/soc/qcom/lpass.h
@@ -82,7 +82,7 @@ struct lpass_variant {
 	 **/
 	u32	dmactl_audif_start;
 	u32	wrdma_channel_start;
-	/* SOC specific intialization like clocks */
+	/* SOC specific initialization like clocks */
 	int (*init)(struct platform_device *pdev);
 	int (*exit)(struct platform_device *pdev);
 	int (*alloc_dma_channel)(struct lpass_data *data, int direction);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a110d39..6dca408 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3041,7 +3041,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component,
 	unsigned int i;
 	int ret;
 
-	dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count);
+	dev_dbg(dev, "ASoC: dai register %s #%zu\n", dev_name(dev), count);
 
 	component->dai_drv = dai_drv;
 
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index aff3d81..3e9b1c0 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -344,7 +344,7 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg,
 	return 0;
 }
 
-/* pass DAI configurations to component driver for extra intialization */
+/* pass DAI configurations to component driver for extra initialization */
 static int soc_tplg_dai_load(struct soc_tplg *tplg,
 	struct snd_soc_dai_driver *dai_drv)
 {
@@ -354,7 +354,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg,
 	return 0;
 }
 
-/* pass link configurations to component driver for extra intialization */
+/* pass link configurations to component driver for extra initialization */
 static int soc_tplg_dai_link_load(struct soc_tplg *tplg,
 	struct snd_soc_dai_link *link)
 {