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From 0d8aa2ccb2f21c79bc9d4dceab0c6f99ff20bae1 Mon Sep 17 00:00:00 2001
From: Randy Dunlap <rdunlap@infradead.org>
Date: Fri, 7 Aug 2020 18:22:09 -0700
Subject: [PATCH] ASoC: various vendors: delete repeated words in comments
References: jsc#SLE-16518
Patch-mainline: v5.10-rc1
Git-commit: 0d8aa2ccb2f21c79bc9d4dceab0c6f99ff20bae1

Drop the repeated words {related, we, is, the} in comments.

Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20200808012209.10880-1-rdunlap@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Acked-by: Takashi Iwai <tiwai@suse.de>

---
 sound/soc/fsl/fsl_dma.c             | 2 +-
 sound/soc/intel/skylake/skl-sst.c   | 2 +-
 sound/soc/meson/axg-tdm-formatter.c | 2 +-
 sound/soc/sprd/sprd-pcm-compress.c  | 2 +-
 sound/soc/sunxi/sun4i-codec.c       | 2 +-
 sound/soc/ti/davinci-mcasp.c        | 2 +-
 6 files changed, 6 insertions(+), 6 deletions(-)

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index be021250d6e9..e0c39c5f4854 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 61a8e4756a2b..00a97cea58b4 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 f7e8e9da68a0..cab7fa2851aa 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 749dcb7b993b..6507c03cc80e 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 2af6404dbd62..6c13cc84b3fb 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 617440767c45..3ffdd0f6292a 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;
-- 
2.16.4