ALSA: pci: delete repeated words in comments

[ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ]

Drop duplicated words in sound/pci/.
{and, the, at}

Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>

diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 5b888b795f7ee85ec74b5fdf0d4581b690d16677..c07a9e735733a6e1899b6d19093b8767ed06c981 100644 (file)
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -766,7 +766,7 @@ 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 715ead59613dad32a53f565a2943ecac94fd4267..0bef823c5f61fd0a522ca0f5c66774c1efd303bf 100644 (file)
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1716,7 +1716,7 @@ 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 801abf0fc98b311ffa19c6b32b409a192eae5d55..103011e7285a3328148d31332e4b68e583506c8a 100644 (file)
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3420,7 +3420,7 @@ 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 6815f9dc8545d42ddbe25375ae3a08f1d976ddaf..e1750bdbe51f6ee16d0d895115a216e2054546b8 100644 (file)
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -813,7 +813,7 @@ 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 4b9300babc7d094cc20ef0161bd9679f6d31e6b0..bfd3fe5eff31c4985b20e4b25dfdc40405d9e28e 100644 (file)
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -832,7 +832,7 @@ 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 98f8ac6587962489bd929758bd32c1d79114ad5a..243f757da3edb2cd4a2fcadd32f629cc36a9ee5e 100644 (file)
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -32,7 +32,7 @@
  *               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 c3f8721624cd4ac63941e1d0259cdd831ff0c16f..b90421a1d909ac8fb7a4afce5d31d0c210284cfd 100644 (file)
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -29,7 +29,7 @@
  *   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
  *