Merge commit 'v2.6.39-rc3' into for-2.6.39

38 files changed, 143 insertions, 90 deletions

diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 4382d0fa6b9a..d8f6fd65ebbb 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -29,7 +29,7 @@
  *      PM support
  *      MIDI support
  *      Game Port support
- *      SG DMA support (this will need *alot* of work)
+ *      SG DMA support (this will need *a lot* of work)
  */
 
 #include <linux/init.h>
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index f53a31e939c1..f8ccc9677c6f 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -963,7 +963,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream)
 
         /*? also check ASI5000 samplerate source
             If external, only support external rate.
-            If internal and other stream playing, cant switch
+            If internal and other stream playing, can't switch
         */
 
         init_timer(&dpcm->timer);
diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index 6fc025c448de..255429c32c1c 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -725,7 +725,7 @@ enum HPI_AESEBU_ERRORS {
 #define HPI_PAD_TITLE_LEN               64
 /** The text string containing the comment. */
 #define HPI_PAD_COMMENT_LEN             256
-/** The PTY when the tuner has not recieved any PTY. */
+/** The PTY when the tuner has not received any PTY. */
 #define HPI_PAD_PROGRAM_TYPE_INVALID    0xffff
 /** \} */
 
diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c
index 3e3c2ef6efd8..8c8aac4c567e 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -423,7 +423,7 @@ static void subsys_create_adapter(struct hpi_message *phm,
 
         ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL);
         if (!ao.priv) {
-                HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n");
+                HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n");
                 phr->error = HPI_ERROR_MEMORY_ALLOC;
                 return;
         }
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 620525bdac59..22e9f08dea6d 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -466,7 +466,7 @@ static void subsys_create_adapter(struct hpi_message *phm,
 
         ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL);
         if (!ao.priv) {
-                HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n");
+                HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n");
                 phr->error = HPI_ERROR_MEMORY_ALLOC;
                 return;
         }
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index af678be0aa15..3b9fd115da36 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -607,7 +607,7 @@ struct hpi_data_compat32 {
 #endif
 
 struct hpi_buffer {
-  /** placehoder for backward compatability (see dwBufferSize) */
+  /** placehoder for backward compatibility (see dwBufferSize) */
         struct hpi_msg_format reserved;
         u32 command; /**< HPI_BUFFER_CMD_xxx*/
         u32 pci_address; /**< PCI physical address of buffer for DSP DMA */
diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c
index bcbdf30a6aa0..360028b9abf5 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -722,7 +722,7 @@ static u16 HPIMSGX__init(struct hpi_message *phm,
                 return phr->error;
         }
         if (hr.error == 0) {
-                /* the adapter was created succesfully
+                /* the adapter was created successfully
                    save the mapping for future use */
                 hpi_entry_points[hr.u.s.adapter_index] = entry_point_func;
                 /* prepare adapter (pre-open streams etc.) */
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index ecb8f4daf408..02f6e08f7592 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -104,7 +104,7 @@
 #define MIX_PLAYB(x) (vortex->mixplayb[x])
 #define MIX_SPDIF(x) (vortex->mixspdif[x])
 
-#define NR_WTPB 0x20            /* WT channels per eahc bank. */
+#define NR_WTPB 0x20            /* WT channels per each bank. */
 
 /* Structs */
 typedef struct {
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index f4aa8ff6f5f9..9ae8b3b17651 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -53,7 +53,7 @@ a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack,
 }
 
 #endif
-/* Atmospheric absorbtion. */
+/* Atmospheric absorption. */
 
 static void
 a3dsrc_SetAtmosTarget(a3dsrc_t * a, short aa, short b, short c, short d,
@@ -835,7 +835,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol,
                 params[i] = ucontrol->value.integer.value[i];
         /* Translate generic filter params to a3d filter params. */
         vortex_a3d_translate_filter(a->filter, params);
-        /* Atmospheric absorbtion and filtering. */
+        /* Atmospheric absorption and filtering. */
         a3dsrc_SetAtmosTarget(a, a->filter[0],
                               a->filter[1], a->filter[2],
                               a->filter[3], a->filter[4]);
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 5439d662d104..33f0ba5559a7 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -515,7 +515,7 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
                 return -ENODEV;
 
         /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the 
-         * same dma engine. WT uses it own separate dma engine whcih cant capture. */
+         * same dma engine. WT uses it own separate dma engine which can't capture. */
         if (idx == VORTEX_PCM_ADB)
                 nr_capt = nr;
         else
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 5715c4d05573..9b7a6346037a 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -140,7 +140,7 @@
  *  Possible remedies:
  *  - use speaker (amplifier) output instead of headphone output
  *    (in case crackling is due to overloaded output clipping)
- *  - plug card into a different PCI slot, preferrably one that isn't shared
+ *  - plug card into a different PCI slot, preferably one that isn't shared
  *    too much (this helps a lot, but not completely!)
  *  - get rid of PCI VGA card, use AGP instead
  *  - upgrade or downgrade BIOS
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index fc53b9bca26d..e8e8ccc96403 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -51,7 +51,7 @@
  *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
  *
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -175,7 +175,7 @@
 /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers                     */
 /********************************************************************************************************/
                                                                                                                            
-/* Initally all registers from 0x00 to 0x3f have zero contents. */
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
 #define PLAYBACK_LIST_ADDR      0x00            /* Base DMA address of a list of pointers to each period/size */
                                                 /* One list entry: 4 bytes for DMA address, 
                                                  * 4 bytes for period_size << 16.
@@ -223,7 +223,7 @@
  * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
  * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
  * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
- * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red.
+ * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
  * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
  */
 /* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 01b49388fafd..437759239694 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -117,7 +117,7 @@
  *    DAC: Unknown
  *    Trying to handle it like the SB0410.
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 630aa4998189..84f3f92436b5 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -42,7 +42,7 @@
  *  0.0.18
  *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index ba96428c9f4c..c694464b1168 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -42,7 +42,7 @@
  *  0.0.18
  *    Implement support for Line-in capture on SB Live 24bit.
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index b5bb036ef73c..f4e573555da3 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -73,7 +73,7 @@ MODULE_PARM_DESC(mpu_port, "MPU-401 port.");
 module_param_array(fm_port, long, NULL, 0444);
 MODULE_PARM_DESC(fm_port, "FM port.");
 module_param_array(soft_ac3, bool, NULL, 0444);
-MODULE_PARM_DESC(soft_ac3, "Sofware-conversion of raw SPDIF packets (model 033 only).");
+MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only).");
 #ifdef SUPPORT_JOYSTICK
 module_param_array(joystick_port, int, NULL, 0444);
 MODULE_PARM_DESC(joystick_port, "Joystick port address.");
@@ -656,8 +656,8 @@ out:
 }
 
 /*
- * Program pll register bits, I assume that the 8 registers 0xf8 upto 0xff
- * are mapped onto the 8 ADC/DAC sampling frequency which can be choosen
+ * Program pll register bits, I assume that the 8 registers 0xf8 up to 0xff
+ * are mapped onto the 8 ADC/DAC sampling frequency which can be chosen
  * at the register CM_REG_FUNCTRL1 (0x04).
  * Problem: other ways are also possible (any information about that?)
  */
@@ -666,7 +666,7 @@ static void snd_cmipci_set_pll(struct cmipci *cm, unsigned int rate, unsigned in
         unsigned int reg = CM_REG_PLL + slot;
         /*
          * Guess that this programs at reg. 0x04 the pos 15:13/12:10
-         * for DSFC/ASFC (000 upto 111).
+         * for DSFC/ASFC (000 up to 111).
          */
 
         /* FIXME: Init (Do we've to set an other register first before programming?) */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b9321544c31c..13f33c0719d3 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1627,7 +1627,7 @@ static struct ct_atc atc_preset __devinitdata = {
  *  Creates and initializes a hardware manager.
  *
  *  Creates kmallocated ct_atc structure. Initializes hardware.
- *  Returns 0 if suceeds, or negative error code if fails.
+ *  Returns 0 if succeeds, or negative error code if fails.
  */
 
 int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index 0cf400f879f9..a5c957db5cea 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1285,7 +1285,7 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info)
         hw_write_20kx(hw, PTPALX, ptp_phys_low);
         hw_write_20kx(hw, PTPAHX, ptp_phys_high);
         hw_write_20kx(hw, TRNCTL, trnctl);
-        hw_write_20kx(hw, TRNIS, 0x200c01); /* realy needed? */
+        hw_write_20kx(hw, TRNIS, 0x200c01); /* really needed? */
 
         return 0;
 }
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 957a311514c8..c250614dadd0 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -248,7 +248,7 @@ static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr)
 /*
  * map the given memory block on PTB.
  * if the block is already mapped, update the link order.
- * if no empty pages are found, tries to release unsed memory blocks
+ * if no empty pages are found, tries to release unused memory blocks
  * and retry the mapping.
  */
 int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 61b8ab39800f..a81dc44228ea 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -69,7 +69,7 @@
  *    ADC: Philips 1361T (Stereo 24bit)
  *    DAC: CS4382-K (8-channel, 24bit, 192Khz)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h
index 00f4817533b1..4e0ee1a9747a 100644
--- a/sound/pci/emu10k1/p16v.h
+++ b/sound/pci/emu10k1/p16v.h
@@ -59,7 +59,7 @@
  *    ADC: Philips 1361T (Stereo 24bit)
  *    DAC: CS4382-K (8-channel, 24bit, 192Khz)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -86,7 +86,7 @@
  * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters.
  */
 
-/* Initally all registers from 0x00 to 0x3f have zero contents. */
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
 #define PLAYBACK_LIST_ADDR      0x00            /* Base DMA address of a list of pointers to each period/size */
                                                 /* One list entry: 4 bytes for DMA address, 
                                                  * 4 bytes for period_size << 16.
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 537cfba829a5..863eafea691f 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -229,6 +229,7 @@ MODULE_PARM_DESC(lineio, "Line In to Rear Out (0 = auto, 1 = force).");
 #define ES_REG_1371_CODEC 0x14  /* W/R: Codec Read/Write register address */
 #define   ES_1371_CODEC_RDY        (1<<31)      /* codec ready */
 #define   ES_1371_CODEC_WIP        (1<<30)      /* codec register access in progress */
+#define   EV_1938_CODEC_MAGIC      (1<<26)
 #define   ES_1371_CODEC_PIRD       (1<<23)      /* codec read/write select register */
 #define   ES_1371_CODEC_WRITE(a,d) ((((a)&0x7f)<<16)|(((d)&0xffff)<<0))
 #define   ES_1371_CODEC_READS(a)   ((((a)&0x7f)<<16)|ES_1371_CODEC_PIRD)
@@ -603,12 +604,18 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531,
 
 #ifdef CHIP1371
 
+static inline bool is_ev1938(struct ensoniq *ensoniq)
+{
+        return ensoniq->pci->device == 0x8938;
+}
+
 static void snd_es1371_codec_write(struct snd_ac97 *ac97,
                                    unsigned short reg, unsigned short val)
 {
         struct ensoniq *ensoniq = ac97->private_data;
-        unsigned int t, x;
+        unsigned int t, x, flag;
 
+        flag = is_ev1938(ensoniq) ? EV_1938_CODEC_MAGIC : 0;
         mutex_lock(&ensoniq->src_mutex);
         for (t = 0; t < POLL_COUNT; t++) {
                 if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) {
@@ -630,7 +637,8 @@ static void snd_es1371_codec_write(struct snd_ac97 *ac97,
                                     0x00010000)
                                         break;
                         }
-                        outl(ES_1371_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1371_CODEC));
+                        outl(ES_1371_CODEC_WRITE(reg, val) | flag,
+                             ES_REG(ensoniq, 1371_CODEC));
                         /* restore SRC reg */
                         snd_es1371_wait_src_ready(ensoniq);
                         outl(x, ES_REG(ensoniq, 1371_SMPRATE));
@@ -647,8 +655,9 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97,
                                             unsigned short reg)
 {
         struct ensoniq *ensoniq = ac97->private_data;
-        unsigned int t, x, fail = 0;
+        unsigned int t, x, flag, fail = 0;
 
+        flag = is_ev1938(ensoniq) ? EV_1938_CODEC_MAGIC : 0;
       __again:
         mutex_lock(&ensoniq->src_mutex);
         for (t = 0; t < POLL_COUNT; t++) {
@@ -671,7 +680,8 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97,
                                     0x00010000)
                                         break;
                         }
-                        outl(ES_1371_CODEC_READS(reg), ES_REG(ensoniq, 1371_CODEC));
+                        outl(ES_1371_CODEC_READS(reg) | flag,
+                             ES_REG(ensoniq, 1371_CODEC));
                         /* restore SRC reg */
                         snd_es1371_wait_src_ready(ensoniq);
                         outl(x, ES_REG(ensoniq, 1371_SMPRATE));
@@ -683,6 +693,11 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97,
                         /* now wait for the stinkin' data (RDY) */
                         for (t = 0; t < POLL_COUNT; t++) {
                                 if ((x = inl(ES_REG(ensoniq, 1371_CODEC))) & ES_1371_CODEC_RDY) {
+                                        if (is_ev1938(ensoniq)) {
+                                                for (t = 0; t < 100; t++)
+                                                        inl(ES_REG(ensoniq, CONTROL));
+                                                x = inl(ES_REG(ensoniq, 1371_CODEC));
+                                        }
                                         mutex_unlock(&ensoniq->src_mutex);
                                         return ES_1371_CODEC_READ(x);
                                 }
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2c79e96d0324..430f41db6044 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3661,7 +3661,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
  * with the proper parameters for set up.
  * ops.cleanup should be called in hw_free for clean up of streams.
  *
- * This function returns 0 if successfull, or a negative error code.
+ * This function returns 0 if successful, or a negative error code.
  */
 int __devinit snd_hda_build_pcms(struct hda_bus *bus)
 {
@@ -4851,7 +4851,7 @@ EXPORT_SYMBOL_HDA(snd_hda_suspend);
  *
  * Returns 0 if successful.
  *
- * This fucntion is defined only when POWER_SAVE isn't set.
+ * This function is defined only when POWER_SAVE isn't set.
  * In the power-save mode, the codec is resumed dynamically.
  */
 int snd_hda_resume(struct hda_bus *bus)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d08cf31596f3..ad97d937d3a8 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3034,6 +3034,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
         SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
         SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+        SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
+        SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
         SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
         SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
         {}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 251773e45f61..715615a88a8d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1280,6 +1280,39 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
                                              stream_tag, format, substream);
 }
 
+static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec,
+                                                    int channels)
+{
+        unsigned int chanmask;
+        int chan = channels ? (channels - 1) : 1;
+
+        switch (channels) {
+        default:
+        case 0:
+        case 2:
+                chanmask = 0x00;
+                break;
+        case 4:
+                chanmask = 0x08;
+                break;
+        case 6:
+                chanmask = 0x0b;
+                break;
+        case 8:
+                chanmask = 0x13;
+                break;
+        }
+
+        /* Set the audio infoframe channel allocation and checksum fields.  The
+         * channel count is computed implicitly by the hardware. */
+        snd_hda_codec_write(codec, 0x1, 0,
+                        Nv_VERB_SET_Channel_Allocation, chanmask);
+
+        snd_hda_codec_write(codec, 0x1, 0,
+                        Nv_VERB_SET_Info_Frame_Checksum,
+                        (0x71 - chan - chanmask));
+}
+
 static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo,
                                    struct hda_codec *codec,
                                    struct snd_pcm_substream *substream)
@@ -1298,6 +1331,10 @@ static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo,
                                 AC_VERB_SET_STREAM_FORMAT, 0);
         }
 
+        /* The audio hardware sends a channel count of 0x7 (8ch) when all the
+         * streams are disabled. */
+        nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8);
+
         return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
 
@@ -1308,37 +1345,16 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
                                      struct snd_pcm_substream *substream)
 {
         int chs;
-        unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id;
+        unsigned int dataDCC1, dataDCC2, channel_id;
         int i;
 
         mutex_lock(&codec->spdif_mutex);
 
         chs = substream->runtime->channels;
-        chan = chs ? (chs - 1) : 1;
 
-        switch (chs) {
-        default:
-        case 0:
-        case 2:
-                chanmask = 0x00;
-                break;
-        case 4:
-                chanmask = 0x08;
-                break;
-        case 6:
-                chanmask = 0x0b;
-                break;
-        case 8:
-                chanmask = 0x13;
-                break;
-        }
         dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT;
         dataDCC2 = 0x2;
 
-        /* set the Audio InforFrame Channel Allocation */
-        snd_hda_codec_write(codec, 0x1, 0,
-                        Nv_VERB_SET_Channel_Allocation, chanmask);
-
         /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
         if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
                 snd_hda_codec_write(codec,
@@ -1413,10 +1429,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
                 }
         }
 
-        /* set the Audio Info Frame Checksum */
-        snd_hda_codec_write(codec, 0x1, 0,
-                        Nv_VERB_SET_Info_Frame_Checksum,
-                        (0x71 - chan - chanmask));
+        nvhdmi_8ch_7x_set_info_frame_parameters(codec, chs);
 
         mutex_unlock(&codec->spdif_mutex);
         return 0;
@@ -1512,6 +1525,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec)
         spec->multiout.max_channels = 8;
         spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x;
         codec->patch_ops = nvhdmi_patch_ops_8ch_7x;
+
+        /* Initialize the audio infoframe channel mask and checksum to something
+         * valid */
+        nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8);
+
         return 0;
 }
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0ef0035fe99f..52928d9a72da 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -549,7 +549,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
 
 /*
  * Control the mode of pin widget settings via the mixer.  "pc" is used
- * instead of "%" to avoid consequences of accidently treating the % as
+ * instead of "%" to avoid consequences of accidentally treating the % as
  * being part of a format specifier.  Maximum allowed length of a value is
  * 63 characters plus NULL terminator.
  *
@@ -9836,7 +9836,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 
         SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 
-        SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
+        SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
         SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
         SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
@@ -9863,7 +9863,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
         SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
         SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
-        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
 
         SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
         SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
@@ -10700,6 +10699,7 @@ enum {
         PINFIX_LENOVO_Y530,
         PINFIX_PB_M5210,
         PINFIX_ACER_ASPIRE_7736,
+        PINFIX_GIGABYTE_880GM,
 };
 
 static const struct alc_fixup alc882_fixups[] = {
@@ -10731,6 +10731,13 @@ static const struct alc_fixup alc882_fixups[] = {
                 .type = ALC_FIXUP_SKU,
                 .v.sku = ALC_FIXUP_SKU_IGNORE,
         },
+        [PINFIX_GIGABYTE_880GM] = {
+                .type = ALC_FIXUP_PINS,
+                .v.pins = (const struct alc_pincfg[]) {
+                        { 0x14, 0x1114410 }, /* set as speaker */
+                        { }
+                }
+        },
 };
 
 static struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -10738,6 +10745,7 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
         SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
         SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
+        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM),
         {}
 };
 
@@ -14116,7 +14124,7 @@ static hda_nid_t alc269vb_capsrc_nids[1] = {
 };
 
 static hda_nid_t alc269_adc_candidates[] = {
-        0x08, 0x09, 0x07,
+        0x08, 0x09, 0x07, 0x11,
 };
 
 #define alc269_modes            alc260_modes
@@ -18774,8 +18782,6 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
                       ALC662_3ST_6ch_DIG),
         SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
         SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
-        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
-                      ALC662_3ST_6ch_DIG),
         SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
         SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
         SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
@@ -19449,6 +19455,7 @@ enum {
         ALC662_FIXUP_IDEAPAD,
         ALC272_FIXUP_MARIO,
         ALC662_FIXUP_CZC_P10T,
+        ALC662_FIXUP_GIGABYTE,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -19477,12 +19484,20 @@ static const struct alc_fixup alc662_fixups[] = {
                         {}
                 }
         },
+        [ALC662_FIXUP_GIGABYTE] = {
+                .type = ALC_FIXUP_PINS,
+                .v.pins = (const struct alc_pincfg[]) {
+                        { 0x14, 0x1114410 }, /* set as speaker */
+                        { }
+                }
+        },
 };
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
         SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
         SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
+        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE),
         SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
         SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
         SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 05fcd60cc46f..94d19c03a7f4 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2475,7 +2475,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
  
         spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0;
 
-        /* check to be sure that the ports are upto date with
+        /* check to be sure that the ports are up to date with
          * switch changes
          */
         stac_issue_unsol_event(codec, nid);
@@ -3408,6 +3408,9 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
         hda_nid_t conn[HDA_MAX_NUM_INPUTS];
         int i, nums;
 
+        if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST))
+                return -1;
+
         nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
         for (i = 0; i < nums; i++)
                 if (conn[i] == nid)
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 2f6252266a02..3e4f8c12ffce 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -148,7 +148,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg,
         udelay(100);
         /*
          * send device address, command and value,
-         * skipping ack cycles inbetween
+         * skipping ack cycles in between
          */
         for (j = 0; j < 3; j++) {
                 switch (j) {
@@ -2143,7 +2143,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice)
                 ice->num_total_adcs = 2;
         }
 
-        /* to remeber the register values of CS8415 */
+        /* to remember the register values of CS8415 */
         ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
         if (!ice->akm)
                 return -ENOMEM;
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 4fc6d8bc637e..f4594d76b6ea 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2755,7 +2755,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
                         return err;
                 }
                 if (c->mpu401_1_name)
-                        /*  Prefered name available in card_info */
+                        /*  Preferred name available in card_info */
                         snprintf(ice->rmidi[0]->name,
                                  sizeof(ice->rmidi[0]->name),
                                  "%s %d", c->mpu401_1_name, card->number);
@@ -2772,7 +2772,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
                                 return err;
                         }
                         if (c->mpu401_2_name)
-                                /*  Prefered name available in card_info */
+                                /*  Preferred name available in card_info */
                                 snprintf(ice->rmidi[1]->name,
                                          sizeof(ice->rmidi[1]->name),
                                          "%s %d", c->mpu401_2_name,
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index cdb873f5da50..92c1160d7ab5 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -768,7 +768,7 @@ static int __devinit pontis_init(struct snd_ice1712 *ice)
         ice->num_total_dacs = 2;
         ice->num_total_adcs = 2;
 
-        /* to remeber the register values */
+        /* to remember the register values */
         ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
         if (! ice->akm)
                 return -ENOMEM;
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 6a9fee3ee78f..764cc93dbca4 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1046,7 +1046,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice)
         * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
         */
         ice->gpio.saved[0] = 0;
-        /* to remeber the register values */
+        /* to remember the register values */
 
         ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
         if (! ice->akm)
@@ -1128,7 +1128,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
         * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
         */
         ice->gpio.saved[0] = 0;
-        /* to remeber the register values */
+        /* to remember the register values */
 
         ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
         if (! ice->akm)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 629a5494347a..6c896dbfd796 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -534,7 +534,7 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code
                 udelay(10);
         } while (time--);
 
-        /* access to some forbidden (non existant) ac97 registers will not
+        /* access to some forbidden (non existent) ac97 registers will not
          * reset the semaphore. So even if you don't get the semaphore, still
          * continue the access. We don't need the semaphore anyway. */
         snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 2ae8d29500a8..27709f0cd2a6 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -331,7 +331,7 @@ static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int co
                 udelay(10);
         } while (time--);
 
-        /* access to some forbidden (non existant) ac97 registers will not
+        /* access to some forbidden (non existent) ac97 registers will not
          * reset the semaphore. So even if you don't get the semaphore, still
          * continue the access. We don't need the semaphore anyway. */
         snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index d3350f383966..3df0f530f67c 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -265,7 +265,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int
         if (! timeout) {
                 /* error - no ack */
                 mutex_unlock(&mgr->msg_mutex);
-                snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame);
+                snd_printk(KERN_ERR "error: no response on msg %x\n", msg_frame);
                 return -EIO;
         }
 
@@ -278,7 +278,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int
         err = get_msg(mgr, &resp, msg_frame);
 
         if( request->message_id != resp.message_id )
-                snd_printk(KERN_ERR "REPONSE ERROR!\n");
+                snd_printk(KERN_ERR "RESPONSE ERROR!\n");
 
         mutex_unlock(&mgr->msg_mutex);
         return err;
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 833e7180ad2d..304411c1fe4b 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -1042,11 +1042,11 @@ void pcxhr_msg_tasklet(unsigned long arg)
         int i, j;
 
         if (mgr->src_it_dsp & PCXHR_IRQ_FREQ_CHANGE)
-                snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occured\n");
+                snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occurred\n");
         if (mgr->src_it_dsp & PCXHR_IRQ_TIME_CODE)
-                snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n");
+                snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occurred\n");
         if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY)
-                snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n");
+                snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occurred\n");
         if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) {
                 /* clear events FREQ_CHANGE and TIME_CODE */
                 pcxhr_init_rmh(prmh, CMD_TEST_IT);
@@ -1055,7 +1055,7 @@ void pcxhr_msg_tasklet(unsigned long arg)
                             err, prmh->stat[0]);
         }
         if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) {
-                snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n");
+                snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occurred\n");
 
                 pcxhr_init_rmh(prmh, CMD_ASYNC);
                 prmh->cmd[0] |= 1;      /* add SEL_ASYNC_EVENTS */
@@ -1233,7 +1233,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
         reg = PCXHR_INPL(mgr, PCXHR_PLX_L2PCIDB);
         PCXHR_OUTPL(mgr, PCXHR_PLX_L2PCIDB, reg);
 
-        /* timer irq occured */
+        /* timer irq occurred */
         if (reg & PCXHR_IRQ_TIMER) {
                 int timer_toggle = reg & PCXHR_IRQ_TIMER;
                 /* is a 24 bit counter */
@@ -1288,7 +1288,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
         if (reg & PCXHR_IRQ_MASK) {
                 if (reg & PCXHR_IRQ_ASYNC) {
                         /* as we didn't request any async notifications,
-                         * some kind of xrun error will probably occured
+                         * some kind of xrun error will probably occurred
                          */
                         /* better resynchronize all streams next interrupt : */
                         mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index d5f5b440fc40..9ff247fc8871 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -150,7 +150,7 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
 #define RME96_RCR_BITPOS_F1 28
 #define RME96_RCR_BITPOS_F2 29
 
-/* Additonal register bits */
+/* Additional register bits */
 #define RME96_AR_WSEL       (1 << 0)
 #define RME96_AR_ANALOG     (1 << 1)
 #define RME96_AR_FREQPAD_0  (1 << 2)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index a323eafb9e03..949691a876d3 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -391,7 +391,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 
 /* Status2 Register bits */ /* MADI ONLY */
 
-#define HDSPM_version0 (1<<0)   /* not realy defined but I guess */
+#define HDSPM_version0 (1<<0)   /* not really defined but I guess */
 #define HDSPM_version1 (1<<1)   /* in former cards it was ??? */
 #define HDSPM_version2 (1<<2)
 
@@ -936,7 +936,7 @@ struct hdspm {
         struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS];
         /* but input to much, so not used */
         struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS];
-        /* full mixer accessable over mixer ioctl or hwdep-device */
+        /* full mixer accessible over mixer ioctl or hwdep-device */
         struct hdspm_mixer *mixer;
 
         struct hdspm_tco *tco;  /* NULL if no TCO detected */
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 1b8f6742b5fa..2b5c7a95ae1f 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -308,7 +308,7 @@ static irqreturn_t sis_interrupt(int irq, void *dev)
         u32 intr, status;
 
         /* We only use the DMA interrupts, and we don't enable any other
-         * source of interrupts. But, it is possible to see an interupt
+         * source of interrupts. But, it is possible to see an interrupt
          * status that didn't actually interrupt us, so eliminate anything
          * we're not expecting to avoid falsely claiming an IRQ, and an
          * ensuing endless loop.
@@ -773,7 +773,7 @@ static void sis_prepare_timing_voice(struct voice *voice,
                 vperiod = 0;
         }
 
-        /* The interrupt handler implements the timing syncronization, so
+        /* The interrupt handler implements the timing synchronization, so
          * setup its state.
          */
         timing->flags |= VOICE_SYNC_TIMING;
@@ -1139,7 +1139,7 @@ static int sis_chip_init(struct sis7019 *sis)
          */
         outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR);
 
-        /* Reset the syncronization groups for all of the channels
+        /* Reset the synchronization groups for all of the channels
          * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc.
          * we'll need to change how we handle these. Until then, we just
          * assign sub-mixer 0 to all playback channels, and avoid any