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| author | Clément Bœsch <u@pkh.me> | 2016-06-21 21:55:20 +0200 |
|---|---|---|
| committer | Clément Bœsch <u@pkh.me> | 2016-06-21 21:55:34 +0200 |
| commit | 8ef57a0d6154119e1a616dd8c29e8c32e35808a0 (patch) | |
| tree | 26c51bc5d99260b44ba3a2585091ca764559f939 /libavformat/dv.c | |
| parent | 373b82066cd4d0c7f42af9b03e8cdc1085e1a6e5 (diff) | |
| parent | 41ed7ab45fc693f7d7fc35664c0233f4c32d69bb (diff) | |
Merge commit '41ed7ab45fc693f7d7fc35664c0233f4c32d69bb'
* commit '41ed7ab45fc693f7d7fc35664c0233f4c32d69bb':
cosmetics: Fix spelling mistakes
Merged-by: Clément Bœsch <u@pkh.me>
Diffstat (limited to 'libavformat/dv.c')
| -rw-r--r-- | libavformat/dv.c | 16 |
1 files changed, 8 insertions, 8 deletions
diff --git a/libavformat/dv.c b/libavformat/dv.c index c689d9b100..89a9e10c14 100644 --- a/libavformat/dv.c +++ b/libavformat/dv.c @@ -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, |