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spelling cosmetics

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Originally committed as revision 15518 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Diego Biurrun 2008-10-02 16:28:58 +00:00
parent fb65d2ca84
commit 910f02a054
10 changed files with 40 additions and 40 deletions
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2
doc/ffmpeg-doc.texi
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@ -894,7 +894,7 @@ motion estimation completely (you have only I-frames, which means it
is about as good as JPEG compression).
@item To have very low audio bitrates, reduce the sampling frequency
(down to 22050 kHz for MPEG audio, 22050 or 11025 for AC3).
(down to 22050kHz for MPEG audio, 22050 or 11025 for AC-3).
@item To have a constant quality (but a variable bitrate), use the option
'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst

6
libavcodec/dv.c
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@ -284,7 +284,7 @@ static inline int put_bits_left(PutBitContext* s)
return (s->buf_end - s->buf) * 8 - put_bits_count(s);
}
/* decode ac coefs */
/* decode ac coefficients */
static void dv_decode_ac(GetBitContext *gb, BlockInfo *mb, DCTELEM *block)
{
int last_index = gb->size_in_bits;
@ -663,7 +663,7 @@ static av_always_inline void dv_set_class_number(DCTELEM* blk, EncBlockInfo* bi,
method suggested in SMPTE 314M Table 22, and an improved
method. The SMPTE method is very conservative; it assigns class
3 (i.e. severe quantization) to any block where the largest AC
component is greater than 36. ffmpeg's DV encoder tracks AC bit
component is greater than 36. FFmpeg's DV encoder tracks AC bit
consumption precisely, so there is no need to bias most blocks
towards strongly lossy compression. Instead, we assign class 2
to most blocks, and use class 3 only when strictly necessary
@ -671,7 +671,7 @@ static av_always_inline void dv_set_class_number(DCTELEM* blk, EncBlockInfo* bi,
#if 0 /* SMPTE spec method */
static const int classes[] = {12, 24, 36, 0xffff};
#else /* improved ffmpeg method */
#else /* improved FFmpeg method */
static const int classes[] = {-1, -1, 255, 0xffff};
#endif
int max=classes[0];

24
libavcodec/dvdata.h
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@ -53,8 +53,8 @@ typedef struct DVprofile {
int bpm; /* blocks per macroblock */
const uint8_t *block_sizes; /* AC block sizes, in bits */
int audio_stride; /* size of audio_shuffle table */
int audio_min_samples[3];/* min ammount of audio samples */
/* for 48Khz, 44.1Khz and 32Khz */
int audio_min_samples[3];/* min amount of audio samples */
/* for 48kHz, 44.1kHz and 32kHz */
int audio_samples_dist[5];/* how many samples are supposed to be */
/* in each frame in a 5 frames window */
const uint8_t (*audio_shuffle)[9]; /* PCM shuffling table */
@ -323,7 +323,7 @@ static const uint8_t dv100_qstep[16] = {
2, 3, 4, 5, 6, 7, 8, 16, 18, 20, 22, 24, 28, 52
};
/* NOTE: I prefer hardcoding the positioning of dv blocks, it is
/* NOTE: I prefer hardcoding the positioning of DV blocks, it is
simpler :-) */
static const uint16_t dv_place_420[1620] = {
@ -6175,7 +6175,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 6,
.block_sizes = block_sizes_dv2550,
.audio_stride = 90,
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
.audio_shuffle = dv_audio_shuffle525,
},
@ -6195,7 +6195,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 6,
.block_sizes = block_sizes_dv2550,
.audio_stride = 108,
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
.audio_shuffle = dv_audio_shuffle625,
},
@ -6215,7 +6215,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 6,
.block_sizes = block_sizes_dv2550,
.audio_stride = 108,
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
.audio_shuffle = dv_audio_shuffle625,
},
@ -6235,7 +6235,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 6,
.block_sizes = block_sizes_dv2550,
.audio_stride = 90,
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
.audio_shuffle = dv_audio_shuffle525,
},
@ -6255,7 +6255,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 6,
.block_sizes = block_sizes_dv2550,
.audio_stride = 108,
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
.audio_shuffle = dv_audio_shuffle625,
},
@ -6275,7 +6275,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 8,
.block_sizes = block_sizes_dv100,
.audio_stride = 90,
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
.audio_shuffle = dv_audio_shuffle525,
},
@ -6295,7 +6295,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 8,
.block_sizes = block_sizes_dv100,
.audio_stride = 108,
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
.audio_shuffle = dv_audio_shuffle625,
},
@ -6315,7 +6315,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 8,
.block_sizes = block_sizes_dv100,
.audio_stride = 90,
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
.audio_shuffle = dv_audio_shuffle525,
},
@ -6335,7 +6335,7 @@ static const DVprofile dv_profiles[] = {
.bpm = 8,
.block_sizes = block_sizes_dv100,
.audio_stride = 90,
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
.audio_shuffle = dv_audio_shuffle525,
}

10
libavformat/dv.c
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@ -112,7 +112,7 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
return 0;
smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
if (quant > 1)
@ -146,7 +146,7 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
continue;
pcm[of*2] = frame[d+1]; // FIXME: maybe we have to admit
pcm[of*2+1] = frame[d]; // that DV is a big endian PCM
pcm[of*2+1] = frame[d]; // that DV is a big-endian PCM
if (pcm[of*2+1] == 0x80 && pcm[of*2] == 0x00)
pcm[of*2+1] = 0;
} else { /* 12bit quantization */
@ -162,11 +162,11 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
continue;
pcm[of*2] = lc & 0xff; // FIXME: maybe we have to admit
pcm[of*2+1] = lc >> 8; // that DV is a big endian PCM
pcm[of*2+1] = lc >> 8; // that DV is a big-endian PCM
of = sys->audio_shuffle[i%half_ch+half_ch][j] +
(d - 8)/3 * sys->audio_stride;
pcm[of*2] = rc & 0xff; // FIXME: maybe we have to admit
pcm[of*2+1] = rc >> 8; // that DV is a big endian PCM
pcm[of*2+1] = rc >> 8; // that DV is a big-endian PCM
++d;
}
}
@ -196,7 +196,7 @@ static int dv_extract_audio_info(DVDemuxContext* c, 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 - 32 kHz */
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 */

8
libavformat/dvenc.c
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@ -38,7 +38,7 @@ struct DVMuxContext {
const DVprofile* sys; /* Current DV profile. E.g.: 525/60, 625/50 */
int n_ast; /* Number of stereo audio streams (up to 2) */
AVStream *ast[2]; /* Stereo audio streams */
AVFifoBuffer audio_data[2]; /* Fifo for storing excessive amounts of PCM */
AVFifoBuffer audio_data[2]; /* FIFO for storing excessive amounts of PCM */
int frames; /* Number of a current frame */
time_t start_time; /* Start time of recording */
int has_audio; /* frame under contruction has audio */
@ -117,7 +117,7 @@ static int dv_write_pack(enum dv_pack_type pack_id, DVMuxContext *c, uint8_t* bu
(c->sys->n_difchan & 2); /* definition: 0 -- 25Mbps, 2 -- 50Mbps */
buf[4] = (1 << 7) | /* emphasis: 1 -- off */
(0 << 6) | /* emphasis time constant: 0 -- reserved */
(0 << 3) | /* frequency: 0 -- 48Khz, 1 -- 44,1Khz, 2 -- 32Khz */
(0 << 3) | /* frequency: 0 -- 48kHz, 1 -- 44,1kHz, 2 -- 32kHz */
0; /* quantization: 0 -- 16bit linear, 1 -- 12bit nonlinear */
va_end(ap);
break;
@ -190,7 +190,7 @@ static void dv_inject_audio(DVMuxContext *c, int channel, uint8_t* frame_ptr)
continue;
frame_ptr[d] = av_fifo_peek(&c->audio_data[channel], of*2+1); // FIXME: maybe we have to admit
frame_ptr[d+1] = av_fifo_peek(&c->audio_data[channel], of*2); // that DV is a big endian PCM
frame_ptr[d+1] = av_fifo_peek(&c->audio_data[channel], of*2); // that DV is a big-endian PCM
}
frame_ptr += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
}
@ -365,7 +365,7 @@ static int dv_write_header(AVFormatContext *s)
if (!dv_init_mux(s)) {
av_log(s, AV_LOG_ERROR, "Can't initialize DV format!\n"
"Make sure that you supply exactly two streams:\n"
" video: 25fps or 29.97fps, audio: 2ch/48Khz/PCM\n"
" video: 25fps or 29.97fps, audio: 2ch/48kHz/PCM\n"
" (50Mbps allows an optional second audio stream)\n");
return -1;
}

2
libavformat/rtp_mpv.c
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@ -104,7 +104,7 @@ void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
memcpy(q, buf1, len);
q += len;
/* 90 KHz time stamp */
/* 90kHz time stamp */
s->timestamp = s->cur_timestamp;
ff_rtp_send_data(s1, s->buf, q - s->buf, (len == size));