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-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c2
-rw-r--r--sound/soc/codecs/alc5623.c2
-rw-r--r--sound/soc/codecs/lm4857.c2
-rw-r--r--sound/soc/codecs/sn95031.c2
-rw-r--r--sound/soc/codecs/tlv320aic26.h4
-rw-r--r--sound/soc/codecs/tlv320aic3x.c2
-rw-r--r--sound/soc/codecs/tlv320dac33.c2
-rw-r--r--sound/soc/codecs/twl4030.c6
-rw-r--r--sound/soc/codecs/wm8580.c2
-rw-r--r--sound/soc/codecs/wm8753.c2
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/codecs/wm8955.c2
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/codecs/wm8991.c2
-rw-r--r--sound/soc/codecs/wm8993.c2
-rw-r--r--sound/soc/codecs/wm9081.c4
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/kirkwood/kirkwood-dma.c4
-rw-r--r--sound/soc/mid-x86/sst_platform.c4
-rw-r--r--sound/soc/omap/ams-delta.c6
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c4
-rw-r--r--sound/soc/sh/fsi.c1
-rw-r--r--sound/soc/soc-jack.c2
23 files changed, 33 insertions, 30 deletions
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 5d230cee3fa7..7fbfa051f6e1 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -672,7 +672,7 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
/* re-enable interrupts */
ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
- /* Re-enable recieve and transmit as appropriate */
+ /* Re-enable receive and transmit as appropriate */
cr = 0;
cr |=
(ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 4f377c9e868d..eecffb548947 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -481,7 +481,7 @@ struct _pll_div {
};
/* Note : pll code from original alc5623 driver. Not sure of how good it is */
-/* usefull only for master mode */
+/* useful only for master mode */
static const struct _pll_div codec_master_pll_div[] = {
{ 2048000, 8192000, 0x0ea0},
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 72de47e5d040..2c2a681da0d7 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -161,7 +161,7 @@ static const struct snd_kcontrol_new lm4857_controls[] = {
lm4857_get_mode, lm4857_set_mode),
};
-/* There is a demux inbetween the the input signal and the output signals.
+/* There is a demux between the input signal and the output signals.
* Currently there is no easy way to model it in ASoC and since it does not make
* much of a difference in practice simply connect the input direclty to the
* outputs. */
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index f70977d7dbe6..84ffdebb8a8b 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -26,7 +26,9 @@
#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
#include <linux/platform_device.h>
+#include <linux/delay.h>
#include <linux/slab.h>
+
#include <asm/intel_scu_ipc.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
index 62b1f2261429..67f19c3bebe6 100644
--- a/sound/soc/codecs/tlv320aic26.h
+++ b/sound/soc/codecs/tlv320aic26.h
@@ -14,14 +14,14 @@
#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset)
#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0)
-/* Page 0: Auxillary data registers */
+/* Page 0: Auxiliary data registers */
#define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05)
#define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06)
#define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07)
#define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09)
#define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A)
-/* Page 1: Auxillary control registers */
+/* Page 1: Auxiliary control registers */
#define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00)
#define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01)
#define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 3bedab26892f..6c43c13f0430 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -884,7 +884,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
if (bypass_pll)
return 0;
- /* Use PLL, compute apropriate setup for j, d, r and p, the closest
+ /* Use PLL, compute appropriate setup for j, d, r and p, the closest
* one wins the game. Try with d==0 first, next with d!=0.
* Constraints for j are according to the datasheet.
* The sysclk is divided by 1000 to prevent integer overflows.
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index eb1a0b4e09b6..082e9d51963f 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1027,7 +1027,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
/*
* For FIFO bypass mode:
* Enable the FIFO bypass (Disable the FIFO use)
- * Set the BCLK as continous
+ * Set the BCLK as continuous
*/
fifoctrl_a |= DAC33_FBYPAS;
aictrl_b |= DAC33_BCLKON;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 8512800f6326..575238d68e5e 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -281,7 +281,7 @@ static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
i, val, twl4030_reg[i]);
}
}
- dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+ dev_dbg(codec->dev, "Found %d non-matching registers. %s\n",
difference, difference ? "Not OK" : "OK");
}
@@ -2018,7 +2018,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
u8 mode;
/* If the system master clock is not 26MHz, the voice PCM interface is
- * not avilable.
+ * not available.
*/
if (twl4030->sysclk != 26000) {
dev_err(codec->dev, "The board is configured for %u Hz, while"
@@ -2028,7 +2028,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
}
/* If the codec mode is not option2, the voice PCM interface is not
- * avilable.
+ * available.
*/
mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
& TWL4030_OPT_MODE;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 8f6b5ee6645b..4bbc0a79f01e 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -772,7 +772,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
snd_soc_write(codec, WM8580_PWRDN1, reg);
- /* Make VMID high impedence */
+ /* Make VMID high impedance */
reg = snd_soc_read(codec, WM8580_ADC_CONTROL1);
reg &= ~0x100;
snd_soc_write(codec, WM8580_ADC_CONTROL1, reg);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 3f09deea8d9d..ffa2ffe5ec11 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1312,7 +1312,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_FMTBIT_S24_LE)
/*
- * The WM8753 supports upto 4 different and mutually exclusive DAI
+ * The WM8753 supports up to 4 different and mutually exclusive DAI
* configurations. This gives 2 PCM's available for use, hifi and voice.
* NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
* is connected between the wm8753 and a BT codec or GSM modem.
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 443ae580445c..9b3bba4df5b3 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1895,7 +1895,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 5e0214d6293e..3c7198779c31 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -176,7 +176,7 @@ static int wm8995_pll_factors(struct device *dev,
return 0;
}
-/* Lookup table specifiying SRATE (table 25 in datasheet); some of the
+/* Lookup table specifying SRATE (table 25 in datasheet); some of the
* output frequencies have been rounded to the standard frequencies
* they are intended to match where the error is slight. */
static struct {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3b71dd65c966..500011eb8b2b 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3137,7 +3137,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("FLL Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 28fdfd66661d..3c2ee1bb73cd 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -981,7 +981,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
reg = snd_soc_read(codec, WM8991_CLOCKING_2);
snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC);
- /* set up N , fractional mode and pre-divisor if neccessary */
+ /* set up N , fractional mode and pre-divisor if necessary */
snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM |
(pll_div.div2 ? WM8991_PRESCALE : 0));
snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 379fa22c5b6c..056aef904347 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -324,7 +324,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 55cdf2982020..91c6b39de50c 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -305,7 +305,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol,
/*
* Stop any attempts to change speaker mode while the speaker is enabled.
*
- * We also have some special anti-pop controls dependant on speaker
+ * We also have some special anti-pop controls dependent on speaker
* mode which must be changed along with the mode.
*/
static int speaker_mode_put(struct snd_kcontrol *kcontrol,
@@ -456,7 +456,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
pr_debug("Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ /* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index c331d65587d8..5b13feca7537 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -16,7 +16,7 @@
* sane processor vendors have a FIFO per AC97 slot, the i.MX has only
* one FIFO which combines all valid receive slots. We cannot even select
* which slots we want to receive. The WM9712 with which this driver
- * was developped with always sends GPIO status data in slot 12 which
+ * was developed with always sends GPIO status data in slot 12 which
* we receive in our (PCM-) data stream. The only chance we have is to
* manually skip this data in the FIQ handler. With sampling rates different
* from 48000Hz not every frame has valid receive data, so the ratio
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index 0fd6a630db01..e13c6ce46328 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
- /* Ensure that all constraints linked to dma burst are fullfilled */
+ /* Ensure that all constraints linked to dma burst are fulfilled */
err = snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
priv->burst * 2,
@@ -170,7 +170,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
/*
* Enable Error interrupts. We're only ack'ing them but
- * it's usefull for diagnostics
+ * it's useful for diagnostics
*/
writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK);
}
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index d827edb3d544..9765fb81a5e3 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -446,7 +446,7 @@ static int sst_platform_remove(struct platform_device *pdev)
snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai));
snd_soc_unregister_platform(&pdev->dev);
- pr_debug("sst_platform_remove sucess\n");
+ pr_debug("sst_platform_remove success\n");
return 0;
}
@@ -469,7 +469,7 @@ module_init(sst_soc_platform_init);
static void __exit sst_soc_platform_exit(void)
{
platform_driver_unregister(&sst_platform_driver);
- pr_debug("sst_soc_platform_exit sucess\n");
+ pr_debug("sst_soc_platform_exit success\n");
}
module_exit(sst_soc_platform_exit);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 3167be689621..462cbcbea74a 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -248,7 +248,7 @@ static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
*/
/* To actually apply any modem controlled configuration changes to the codec,
- * we must connect codec DAI pins to the modem for a moment. Be carefull not
+ * we must connect codec DAI pins to the modem for a moment. Be careful not
* to interfere with our digital mute function that shares the same hardware. */
static struct timer_list cx81801_timer;
static bool cx81801_cmd_pending;
@@ -402,9 +402,9 @@ static struct tty_ldisc_ops cx81801_ops = {
/*
- * Even if not very usefull, the sound card can still work without any of the
+ * Even if not very useful, the sound card can still work without any of the
* above functonality activated. You can still control its audio input/output
- * constellation and speakerphone gain from userspace by issueing AT commands
+ * constellation and speakerphone gain from userspace by issuing AT commands
* over the modem port.
*/
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 78bfdb3f5d7e..452230975632 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -228,7 +228,7 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
SOC_DAPM_PIN_SWITCH("Handset Mic"),
};
-/* GTA02 specific routes and controlls */
+/* GTA02 specific routes and controls */
#ifdef CONFIG_MACH_NEO1973_GTA02
@@ -372,7 +372,7 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-/* GTA01 specific controlls */
+/* GTA01 specific controls */
#ifdef CONFIG_MACH_NEO1973_GTA01
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 584315895393..23c0e83d4c19 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1330,3 +1330,4 @@ module_exit(fsi_mobile_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_ALIAS("platform:fsi-pcm-audio");
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 6203a72d57af..7c17b98d5846 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -331,7 +331,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
goto err;
if (gpios[i].wake) {
- ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
+ ret = irq_set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
if (ret != 0)
printk(KERN_ERR
"Failed to mark GPIO %d as wake source: %d\n",
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-05 08:35:01 +0000