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authorRandy Dunlap <rdunlap@infradead.org>2020-08-08 03:22:09 +0200
committerMark Brown <broonie@kernel.org>2020-08-18 15:52:32 +0200
commit0d8aa2ccb2f21c79bc9d4dceab0c6f99ff20bae1 (patch)
tree85a187525a8aa258ab5c3216899adf350dadf1bb
parentMerge existing fixes from asoc/for-5.9 (diff)
downloadlinux-0d8aa2ccb2f21c79bc9d4dceab0c6f99ff20bae1.tar.xz
linux-0d8aa2ccb2f21c79bc9d4dceab0c6f99ff20bae1.zip
ASoC: various vendors: delete repeated words in comments
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>
-rw-r--r--sound/soc/fsl/fsl_dma.c2
-rw-r--r--sound/soc/intel/skylake/skl-sst.c2
-rw-r--r--sound/soc/meson/axg-tdm-formatter.c2
-rw-r--r--sound/soc/sprd/sprd-pcm-compress.c2
-rw-r--r--sound/soc/sunxi/sun4i-codec.c2
-rw-r--r--sound/soc/ti/davinci-mcasp.c2
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;