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Merge commit '41ed7ab45fc693f7d7fc35664c0233f4c32d69bb'

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* commit '41ed7ab45fc693f7d7fc35664c0233f4c32d69bb':
  cosmetics: Fix spelling mistakes

Merged-by: Clément Bœsch <u@pkh.me>
This commit is contained in:
Clément Bœsch
2016-06-21 21:55:20 +02:00
parent 373b82066c 41ed7ab45f
commit 8ef57a0d61
320 changed files with 870 additions and 871 deletions
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16
libavformat/dv.c
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@@ -110,11 +110,11 @@ static const int dv_audio_frequency[3] = {
/*
* There's a couple of assumptions being made here:
* 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples.
* 1. By default we silence erroneous (0x8000/16-bit 0x800/12-bit) audio samples.
* We can pass them upwards when libavcodec will be ready to deal with them.
* 2. We don't do software emphasis.
* 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
* are converted into 16bit linear ones.
* 3. Audio is always returned as 16-bit linear samples: 12-bit nonlinear samples
* are converted into 16-bit linear ones.
*/
static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
const AVDVProfile *sys)
@@ -130,7 +130,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
freq = as_pack[4] >> 3 & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
quant = as_pack[4] & 0x07; /* 0 - 16-bit linear, 1 - 12-bit nonlinear */
if (quant > 1)
return -1; /* unsupported quantization */
@@ -161,7 +161,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
for (i = 0; i < sys->difseg_size; i++) {
frame += 6 * 80; /* skip DIF segment header */
if (quant == 1 && i == half_ch) {
/* next stereo channel (12bit mode only) */
/* next stereo channel (12-bit mode only) */
av_assert0(ipcm<4);
pcm = ppcm[ipcm++];
if (!pcm)
@@ -171,7 +171,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
/* for each AV sequence */
for (j = 0; j < 9; j++) {
for (d = 8; d < 80; d += 2) {
if (quant == 0) { /* 16bit quantization */
if (quant == 0) { /* 16-bit quantization */
of = sys->audio_shuffle[i][j] +
(d - 8) / 2 * sys->audio_stride;
if (of * 2 >= size)
@@ -184,7 +184,7 @@ static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,
if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)
pcm[of * 2 + 1] = 0;
} else { /* 12bit quantization */
} else { /* 12-bit quantization */
lc = ((uint16_t)frame[d] << 4) |
((uint16_t)frame[d + 2] >> 4);
rc = ((uint16_t)frame[d + 1] << 4) |
@@ -233,7 +233,7 @@ static int dv_extract_audio_info(DVDemuxContext *c, const uint8_t *frame)
smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
freq = as_pack[4] >> 3 & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
stype = as_pack[3] & 0x1f; /* 0 - 2CH, 2 - 4CH, 3 - 8CH */
quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
quant = as_pack[4] & 0x07; /* 0 - 16-bit linear, 1 - 12-bit nonlinear */
if (freq >= FF_ARRAY_ELEMS(dv_audio_frequency)) {
av_log(c->fctx, AV_LOG_ERROR,