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-rw-r--r--sound/pci/ac97/ac97_patch.c2
-rw-r--r--sound/pci/cs46xx/cs46xx_dsp_task_types.h2
-rw-r--r--sound/pci/hda/patch_ca0132.c2
-rw-r--r--sound/pci/ice1712/wm8766.c2
-rw-r--r--sound/pci/ice1712/wm8776.c2
-rw-r--r--sound/pci/korg1212/korg1212.c4
-rw-r--r--sound/pci/pcxhr/pcxhr_hwdep.c2
-rw-r--r--sound/pcmcia/vx/vxp_ops.c2
-rw-r--r--sound/ppc/snd_ps3.c2
-rw-r--r--sound/soc/amd/acp-pcm-dma.c4
-rw-r--r--sound/soc/codecs/wm_hubs.c2
-rw-r--r--sound/soc/fsl/fsl_asrc.c2
-rw-r--r--sound/soc/qcom/lpass.h2
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/soc-topology.c4
15 files changed, 18 insertions, 18 deletions
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index f4234edb878c..8cf0dc7a07a4 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 5cf920bfda27..be5694718546 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 9ec4dba8a793..07a9deb17477 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 f7ac8d5e862c..27c03e40c9b1 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 ebd2fe4b4a57..553669b103c2 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 565f7f55c3ca..1e25095fd144 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 80633055e17e..a99808ab01fe 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 56aa1ba73ccc..5f97791f00d7 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 b84d7d34f188..cdd44abfc9e0 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 818b052377f3..08b1399d1da2 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -89,7 +89,7 @@ static void acp_reg_write(u32 val, void __iomem *acp_mmio, u32 reg)
writel(val, acp_mmio + (reg * 4));
}
-/* Configure a given dma channel parameters - enable/disble,
+/* Configure a given dma channel parameters - enable/disable,
* number of descriptors, priority
*/
static void config_acp_dma_channel(void __iomem *acp_mmio, u8 ch_num,
@@ -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 624b3b9cb079..63b2745f8169 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 1d82f68305c3..8cfffa70c144 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 924971b6ded5..9b031352ea3c 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 a110d3987d4a..6dca408faae3 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 aff3d8129ac9..3e9b1c0bb1ce 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)
{
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-07 21:09:18 +0000