diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 146e1a3498c7..419da70cd942 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned
 		rate = 48000 / 9;
 
 	/*
-	 *  We can not capture at at rate greater than the Input Rate (48000).
+	 *  We can not capture at a rate greater than the Input Rate (48000).
 	 *  Return an error if an attempt is made to stray outside that limit.
 	 */
 	if (rate > 48000)
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 8d0a3d357345..8ef51a29380a 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1739,7 +1739,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip)
 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
 
 	if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
-		/* remove AsynchFGTxSCB and and PCMSerialInput_II */
+		/* remove AsynchFGTxSCB and PCMSerialInput_II */
 		cs46xx_dsp_disable_spdif_out (chip);
 
 		/* save state */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index f3a6b1d869d8..dbeb62362f1c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3410,7 +3410,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save);
  * @nid: NID to check / update
  *
  * Check whether the given NID is in the amp list.  If it's in the list,
- * check the current AMP status, and update the the power-status according
+ * check the current AMP status, and update the power-status according
  * to the mute status.
  *
  * This function is supposed to be set or called from the check_power_status
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 2609161707a4..97adb7e340f9 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -825,7 +825,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path,
 	}
 }
 
-/* sync power of each widget in the the given path */
+/* sync power of each widget in the given path */
 static hda_nid_t path_power_update(struct hda_codec *codec,
 				   struct nid_path *path,
 				   bool allow_powerdown)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d8168aa2cef3..85c33f528d7b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -845,7 +845,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec,
 	static struct snd_kcontrol_new beep_vol_ctl =
 		HDA_CODEC_VOLUME(NULL, 0, 0, 0);
 
-	/* check for mute support for the the amp */
+	/* check for mute support for the amp */
 	if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
 		const struct snd_kcontrol_new *temp;
 		if (spec->anabeep_nid == nid)
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 3919aed39ca0..5e52086d7b98 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -31,7 +31,7 @@
  *		  Experimentally I found out that only a combination of
  *		  OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 -
  *		  VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct
- *		  sampling rate. That means the the FPGA doubles the
+ *		  sampling rate. That means that the FPGA doubles the
  *		  MCK01 rate.
  *
  *	Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index 4cf3200e988b..df44135e1b0c 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -39,7 +39,7 @@
  *   GPIO 4 <- headphone detect
  *   GPIO 5 -> enable ADC analog circuit for the left channel
  *   GPIO 6 -> enable ADC analog circuit for the right channel
- *   GPIO 7 -> switch green rear output jack between CS4245 and and the first
+ *   GPIO 7 -> switch green rear output jack between CS4245 and the first
  *             channel of CS4361 (mechanical relay)
  *   GPIO 8 -> enable output to speakers
  *