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ASoC: fixes multiple typos in comments, no functional change

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Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This commit is contained in:
Peter Meerwald 2009-07-13 23:05:11 +01:00 committed by Mark Brown
parent 0a0cf58d93
commit 47db8e89ac
1 changed files with 13 additions and 13 deletions
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26
include/sound/soc-dai.h
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@ -27,8 +27,8 @@ struct snd_pcm_substream;
#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */
#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
/* left and right justified also known as MSB and LSB respectively */
@ -38,7 +38,7 @@ struct snd_pcm_substream;
/*
* DAI Clock gating.
*
* DAI bit clocks can be be gated (disabled) when not the DAI is not
* DAI bit clocks can be be gated (disabled) when the DAI is not
* sending or receiving PCM data in a frame. This can be used to save power.
*/
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
@ -51,21 +51,21 @@ struct snd_pcm_substream;
* format.
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */
#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */
#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */
/*
* DAI hardware clock masters.
*
* This is wrt the codec, the inverse is true for the interface
* i.e. if the codec is clk and frm master then the interface is
* i.e. if the codec is clk and FRM master then the interface is
* clk and frame slave.
*/
#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */
#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */
#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */
#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */
#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
@ -116,12 +116,12 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
/*
* Digital Audio Interface.
*
* Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
* operations an capabilities. Codec and platfom drivers will register a this
* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
* operations and capabilities. Codec and platform drivers will register this
* structure for every DAI they have.
*
* This structure covers the clocking, formating and ALSA operations for each
* interface a
* interface.
*/
struct snd_soc_dai_ops {
/*