diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index be021250d6e9b0715ea882a4a8c5c55b1a0bc5e7..e0c39c5f4854226c6a8d0a54cc4b2e5bc054077a 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -154,7 +154,7 @@ static void fsl_dma_abort_stream(struct snd_pcm_substream *substream)
 /**
  * fsl_dma_update_pointers - update LD pointers to point to the next period
  *
- * As each period is completed, this function changes the the link
+ * As each period is completed, this function changes the link
  * descriptor pointers for that period to point to the next period.
  */
 static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private)
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 61a8e4756a2b0e5730a3fe7dd7943447dcc75f46..00a97cea58b4e960021604321081155dab335fe1 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -354,7 +354,7 @@ static int skl_transfer_module(struct sst_dsp *ctx, const void *data,
 	/*
 	 * if bytes_left > 0 then wait for BDL complete interrupt and
 	 * copy the next chunk till bytes_left is 0. if bytes_left is
-	 * is zero, then wait for load module IPC reply
+	 * zero, then wait for load module IPC reply
 	 */
 	while (bytes_left > 0) {
 		curr_pos = size - bytes_left;
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index f7e8e9da68a06bbd9e951be45ea9a12880259d16..cab7fa2851aa848a08aa8ef3804adecc12a2f5a6 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -398,7 +398,7 @@ void axg_tdm_stream_free(struct axg_tdm_stream *ts)
 	/*
 	 * If the list is not empty, it would mean that one of the formatter
 	 * widget is still powered and attached to the interface while we
-	 * we are removing the TDM DAI. It should not be possible
+	 * are removing the TDM DAI. It should not be possible
 	 */
 	WARN_ON(!list_empty(&ts->formatter_list));
 	mutex_destroy(&ts->lock);
diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c
index 749dcb7b993b881045359b26256c4c1768227994..6507c03cc80e8ed2a5b2de46540f2796a5427789 100644
--- a/sound/soc/sprd/sprd-pcm-compress.c
+++ b/sound/soc/sprd/sprd-pcm-compress.c
@@ -559,7 +559,7 @@ static int sprd_platform_compr_copy(struct snd_soc_component *component,
 		} else {
 			/*
 			 * If the data count is larger than the available spaces
-			 * of the the stage 0 IRAM buffer, we should copy one
+			 * of the stage 0 IRAM buffer, we should copy one
 			 * partial data to the stage 0 IRAM buffer, and copy
 			 * the left to the stage 1 DDR buffer.
 			 */
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 2af6404dbd62fdaa740304fb8e193850329e1ad6..6c13cc84b3fb553f2ab98c2bf4fd17951f2704f7 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -335,7 +335,7 @@ static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream,
 
 	/*
 	 * FIXME: Undocumented in the datasheet, but
-	 *        Allwinner's code mentions that it is related
+	 *        Allwinner's code mentions that it is
 	 *        related to microphone gain
 	 */
 	if (of_device_is_compatible(scodec->dev->of_node,
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 617440767c45fd66c0f6e1b113823804a29e44d2..3ffdd0f6292acef95e7b178a1484803dd779fde4 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -633,7 +633,7 @@ static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id,
 		 * right channels), so it has to be divided by number
 		 * of tdm-slots (for I2S - divided by 2).
 		 * Instead of storing this ratio, we calculate a new
-		 * tdm_slot width by dividing the the ratio by the
+		 * tdm_slot width by dividing the ratio by the
 		 * number of configured tdm slots.
 		 */
 		mcasp->slot_width = div / mcasp->tdm_slots;