30 files changed, 74 insertions, 59 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 2a30eae1881c..a54d2a5b28f6 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 00b6d87e7bdb..082e9d51963f 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)
@@ -670,6 +674,7 @@ 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 +682,10 @@ 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 +697,11 @@ 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 +719,14 @@ 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 +779,11 @@ 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)
@@ -1020,7 +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;
@@ -1173,15 +1180,16 @@ 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 +1254,10 @@ 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 */
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/twl6040.c b/sound/soc/codecs/twl6040.c
index 482fcdb59bfa..255901c4460d 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);
 
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/wm8994.c b/sound/soc/codecs/wm8994.c
index 3dc64c8b6a5c..3290333b2bb9 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 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-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 671ef8dd524c..aab7765f401a 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
                 slave_config.direction = DMA_TO_DEVICE;
                 slave_config.dst_addr = dma_params->dma_addr;
                 slave_config.dst_addr_width = buswidth;
-                slave_config.dst_maxburst = dma_params->burstsize;
+                slave_config.dst_maxburst = dma_params->burstsize * buswidth;
         } else {
                 slave_config.direction = DMA_FROM_DEVICE;
                 slave_config.src_addr = dma_params->dma_addr;
                 slave_config.src_addr_width = buswidth;
-                slave_config.src_maxburst = dma_params->burstsize;
+                slave_config.src_maxburst = dma_params->burstsize * buswidth;
         }
 
         ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config);
@@ -303,6 +303,11 @@ static struct snd_soc_platform_driver imx_soc_platform_mx2 = {
 
 static int __devinit imx_soc_platform_probe(struct platform_device *pdev)
 {
+        struct imx_ssi *ssi = platform_get_drvdata(pdev);
+
+        ssi->dma_params_tx.burstsize = 6;
+        ssi->dma_params_rx.burstsize = 4;
+
         return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2);
 }
 
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index bc92ec620004..ac2ded969253 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/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index a4406a134892..dc8a87530e3e 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -234,7 +234,4 @@ void imx_pcm_free(struct snd_pcm *pcm);
  */
 #define IMX_SSI_DMABUF_SIZE     (64 * 1024)
 
-#define DMA_RXFIFO_BURST      0x4
-#define DMA_TXFIFO_BURST      0x6
-
 #endif /* _IMX_SSI_H */
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 ee2c22475a76..b2e9198a983a 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 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/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 784cff5f67e8..9027da466cae 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -310,7 +310,7 @@ static struct snd_soc_dai_link corgi_dai = {
         .cpu_dai_name = "pxa2xx-i2s",
         .codec_dai_name = "wm8731-hifi",
         .platform_name = "pxa-pcm-audio",
-        .codec_name = "wm8731-codec-0.001b",
+        .codec_name = "wm8731-codec.0-001b",
         .init = corgi_wm8731_init,
         .ops = &corgi_ops,
 };
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 02fb66416ddc..2ce0b2d891d5 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -65,6 +65,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
         if (prtd->dma_ch >= 0) {
                 pxa_free_dma(prtd->dma_ch);
                 prtd->dma_ch = -1;
+                prtd->params = NULL;
         }
 
         return 0;
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ac577263b3e3..b6445757fc54 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
         .codec_name = "wm9713-codec",
         .platform_name = "pxa-pcm-audio",
         .cpu_dai_name = "pxa2xx-ac97",
-        .codec_name = "wm9713-hifi",
+        .codec_dai_name = "wm9713-hifi",
         .init = zylonite_wm9713_init,
 },
 {
@@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
         .codec_name = "wm9713-codec",
         .platform_name = "pxa-pcm-audio",
         .cpu_dai_name = "pxa2xx-ac97-aux",
-        .codec_name = "wm9713-aux",
+        .codec_dai_name = "wm9713-aux",
 },
 {
         .name = "WM9713 Voice",
@@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
         .codec_name = "wm9713-codec",
         .platform_name = "pxa-pcm-audio",
         .cpu_dai_name = "pxa-ssp-dai.2",
-        .codec_name = "wm9713-voice",
+        .codec_dai_name = "wm9713-voice",
         .ops = &zylonite_voice_ops,
 },
 };
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/soc-core.c b/sound/soc/soc-core.c
index 4dda58926bc5..b76b74db0968 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -92,8 +92,8 @@ static int min_bytes_needed(unsigned long val)
 static int format_register_str(struct snd_soc_codec *codec,
                                unsigned int reg, char *buf, size_t len)
 {
-        int wordsize = codec->driver->reg_word_size * 2;
-        int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+        int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+        int regsize = codec->driver->reg_word_size * 2;
         int ret;
         char tmpbuf[len + 1];
         char regbuf[regsize + 1];
@@ -132,8 +132,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
         size_t total = 0;
         loff_t p = 0;
 
-        wordsize = codec->driver->reg_word_size * 2;
-        regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+        wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+        regsize = codec->driver->reg_word_size * 2;
 
         len = wordsize + regsize + 2 + 1;
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index fcab80b36a37..fc017c0a7b5d 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",