@@ -766,7 +766,7 @@ static void snd_cs46xx_set_capture_sampl
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)
@@ -1716,7 +1716,7 @@ int cs46xx_iec958_pre_open (struct snd_c
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 */
@@ -3428,7 +3428,7 @@ 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
@@ -813,7 +813,7 @@ static void activate_amp_in(struct hda_c
}
}
-/* 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)
@@ -838,7 +838,7 @@ static int stac_auto_create_beep_ctls(st
static const 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)
@@ -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>
@@ -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
*
Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap@infradead.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: Clemens Ladisch <clemens@ladisch.de> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-)