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 0020fd0efc466b46e4482a106d80c23d33546504..09c547f4cc186fc5d683622f129faa849a4bd8f9 100644 (file)
--- 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 7488e1b7a770732cddd8ffc593359bb2d36ed93b..4e726d39b05d125796ce565f33f89bc70badbbd5 100644 (file)
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1742,7 +1742,7 @@ 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 7d65fe31c825731279790774aa8692ec51b1866d..a56f018d586f585f3bad2ac46b59bf6809841a2f 100644 (file)
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3394,7 +3394,7 @@ 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 28ef409a9e6ae3ecf7e266612d01000f0123bdbf..9dee657ce9e27eda607ee4674f8cf0ecfbae3321 100644 (file)
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -823,7 +823,7 @@ 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 7cd147411b22d2398111e8fd2d9c9fa50a6e9deb..f7896a9ae3d658c93a8ca4df2136b095f08aa855 100644 (file)
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -863,7 +863,7 @@ 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 3919aed39ca03ab3005e0bfec8ffd2e65f20a147..5e52086d7b98637357f6a4354ac472e773e1eb0d 100644 (file)
--- 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 4cf3200e988b03adf40129af71cfe1c9bb51bb69..df44135e1b0c9c03e643052e1c76e566626412b9 100644 (file)
--- 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
  *