From a3b55791b5b2b4b6d31ece9b3975466ec2f07b1b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 18 Mar 2011 15:15:11 +0200 Subject: ASoC: tlv320dac33: Fix inconsistent spinlock usage The lock is used within the interrupt handler. Correct the spinlock usage, and use irqsave/irqrestore flavour of spin_lock/unlock. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 28 ++++++++++++++++------------ 1 file changed, 16 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 71d7be8ac48..93f38b1d624 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -670,6 +670,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; unsigned int delay; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: @@ -677,10 +678,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) DAC33_THRREG(dac33->nsample)); /* Take the timestamps */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); dac33->t_stamp1 = dac33->t_stamp2; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); @@ -692,11 +693,11 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); /* Move back the timestamp with drain time */ dac33->t_stamp1 -= dac33->mode7_us_to_lthr; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(DAC33_MODE7_MARGIN)); @@ -714,13 +715,14 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_NSAMPLE_MSB, DAC33_THRREG(dac33->nsample)); @@ -773,10 +775,11 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) { struct snd_soc_codec *codec = dev; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) @@ -1173,15 +1176,16 @@ static snd_pcm_sframes_t dac33_dai_delay( unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_BYPASS: break; case DAC33_FIFO_MODE1: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; t1 = dac33->t_stamp2; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ @@ -1246,10 +1250,10 @@ static snd_pcm_sframes_t dac33_dai_delay( } break; case DAC33_FIFO_MODE7: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; uthr = dac33->uthr; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ -- cgit v1.2.3 From 56a3536c226f58497a8b1b7959b581983bcf04c9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 24 Mar 2011 08:58:05 +0200 Subject: ASoC: tlv320dac33: Restore L/R DAC power control register Register 0x40, 0x41 need to be restored after power up, since it contains gain related fields, which affects playback volume. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 93f38b1d624..6c3735dcb05 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -324,6 +324,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_write(codec, DAC33_OUT_AMP_CTRL, dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL)); + dac33_write(codec, DAC33_LDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_LDAC_PWR_CTRL)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_RDAC_PWR_CTRL)); } static inline int dac33_read_id(struct snd_soc_codec *codec) -- cgit v1.2.3 From 19aab08d69e05c8d9a9ad315e59dd0493bf86045 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 26 Mar 2011 15:53:58 +0800 Subject: ASoC: twl6040: Return -ENOMEM if create_singlethread_workqueue fails Signed-off-by: Axel Lin Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/twl6040.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 482fcdb59bf..255901c4460 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1629,8 +1629,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->naudint = naudint; priv->workqueue = create_singlethread_workqueue("twl6040-codec"); - if (!priv->workqueue) + if (!priv->workqueue) { + ret = -ENOMEM; goto work_err; + } INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); -- cgit v1.2.3 From 458f7f8f819b63431f1098a086fafce4a503fb77 Mon Sep 17 00:00:00 2001 From: Thomas Gleixner Date: Wed, 30 Mar 2011 00:39:44 +0200 Subject: sound: Fixup the last user of the old irq functions I had seen that before, but now that I removed set_irq_wake it broke. Signed-off-by: Thomas Gleixner --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcab80b36a3..fc017c0a7b5 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", -- cgit v1.2.3 From 438008af63f8fb3a061c31d331e150fba6eaa103 Mon Sep 17 00:00:00 2001 From: Ingo Molnar Date: Thu, 13 Nov 2008 11:51:57 +0100 Subject: sound: Add delay.h to sound/soc/codecs/sn95031.c MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is further fallout from delay.h removal from asm/apic.h and asm/dma.h: ca444564a947: x86: Stop including in two asm header files Which caused this build failure: sound/soc/codecs/sn95031.c: In function ‘sn95031_get_mic_bias’: sound/soc/codecs/sn95031.c:153:2: error: implicit declaration of function ‘msleep’ [-Werror=implicit-function-declaration] Cc: Jean Delvare Cc: James E.J. Bottomley Cc: Geert Uytterhoeven Cc: Stephen Rothwell LKML-Reference: <20110325152014.297890ec@endymion.delvare> Signed-off-by: Ingo Molnar --- sound/soc/codecs/sn95031.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 2a30eae1881..a54d2a5b28f 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 +#include #include + #include #include #include -- cgit v1.2.3 From 25985edcedea6396277003854657b5f3cb31a628 Mon Sep 17 00:00:00 2001 From: Lucas De Marchi Date: Wed, 30 Mar 2011 22:57:33 -0300 Subject: Fix common misspellings Fixes generated by 'codespell' and manually reviewed. Signed-off-by: Lucas De Marchi --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/tlv320aic26.h | 4 ++-- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 6 +++--- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 6 +++--- sound/soc/codecs/wm9081.c | 4 ++-- sound/soc/imx/imx-ssi.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 4 ++-- sound/soc/mid-x86/sst_platform.c | 4 ++-- sound/soc/omap/ams-delta.c | 6 +++--- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- 21 files changed, 32 insertions(+), 32 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 5d230cee3fa..7fbfa051f6e 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 4f377c9e868..eecffb54894 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 72de47e5d04..2c2a681da0d 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/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 62b1f226142..67f19c3bebe 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 3bedab26892..6c43c13f043 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 00b6d87e7bd..f01f1417da4 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1020,7 +1020,7 @@ 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 8512800f632..575238d68e5 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 8f6b5ee6645..4bbc0a79f01 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 3f09deea8d9..ffa2ffe5ec1 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 443ae580445..9b3bba4df5b 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 5e0214d6293..3c7198779c3 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 3b71dd65c96..500011eb8b2 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 28fdfd66661..3c2ee1bb73c 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 379fa22c5b6..056aef90434 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/wm8994.c b/sound/soc/codecs/wm8994.c index 3dc64c8b6a5..3290333b2bb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -82,18 +82,18 @@ struct wm8994_priv { int mbc_ena[3]; - /* Platform dependant DRC configuration */ + /* Platform dependent DRC configuration */ const char **drc_texts; int drc_cfg[WM8994_NUM_DRC]; struct soc_enum drc_enum; - /* Platform dependant ReTune mobile configuration */ + /* Platform dependent ReTune mobile configuration */ int num_retune_mobile_texts; const char **retune_mobile_texts; int retune_mobile_cfg[WM8994_NUM_EQ]; struct soc_enum retune_mobile_enum; - /* Platform dependant MBC configuration */ + /* Platform dependent MBC configuration */ int mbc_cfg; const char **mbc_texts; struct soc_enum mbc_enum; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 55cdf298202..91c6b39de50 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 bc92ec62000..ac2ded96925 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 0fd6a630db0..e13c6ce4632 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 ee2c22475a7..b2e9198a983 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -440,7 +440,7 @@ 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; } @@ -463,7 +463,7 @@ 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 3167be68962..462cbcbea74 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 78bfdb3f5d7..45223097563 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 -- cgit v1.2.3