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    Subject[PATCH 4.19 009/125] ALSA: pci: delete repeated words in comments
    Date
    From: Randy Dunlap <rdunlap@infradead.org>

    [ 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>
    ---
    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(-)

    diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
    index 146e1a3498c73..419da70cd942a 100644
    --- 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 8d0a3d3573457..8ef51a29380af 100644
    --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
    +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
    @@ -1739,7 +1739,7 @@ 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 f3a6b1d869d8a..dbeb62362f1c3 100644
    --- a/sound/pci/hda/hda_codec.c
    +++ b/sound/pci/hda/hda_codec.c
    @@ -3410,7 +3410,7 @@ 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 2609161707a41..97adb7e340f99 100644
    --- a/sound/pci/hda/hda_generic.c
    +++ b/sound/pci/hda/hda_generic.c
    @@ -825,7 +825,7 @@ 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 d8168aa2cef38..85c33f528d7b3 100644
    --- a/sound/pci/hda/patch_sigmatel.c
    +++ b/sound/pci/hda/patch_sigmatel.c
    @@ -845,7 +845,7 @@ 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 3919aed39ca03..5e52086d7b986 100644
    --- 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 4cf3200e988b0..df44135e1b0c9 100644
    --- 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
    *
    --
    2.25.1


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