spelling cosmetics
Originally committed as revision 15518 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Diego Biurrun
parent
fb65d2ca84
commit
910f02a054
@ -894,7 +894,7 @@ motion estimation completely (you have only I-frames, which means it
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is about as good as JPEG compression).
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@item To have very low audio bitrates, reduce the sampling frequency
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(down to 22050 kHz for MPEG audio, 22050 or 11025 for AC3).
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(down to 22050kHz for MPEG audio, 22050 or 11025 for AC-3).
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@item To have a constant quality (but a variable bitrate), use the option
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'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
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@ -284,7 +284,7 @@ static inline int put_bits_left(PutBitContext* s)
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return (s->buf_end - s->buf) * 8 - put_bits_count(s);
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}
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/* decode ac coefs */
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/* decode ac coefficients */
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static void dv_decode_ac(GetBitContext *gb, BlockInfo *mb, DCTELEM *block)
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{
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int last_index = gb->size_in_bits;
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@ -493,7 +493,7 @@ static inline void dv_decode_video_segment(DVVideoContext *s,
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mb_y = v >> 8;
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/* We work with 720p frames split in half. The odd half-frame (chan==2,3) is displaced :-( */
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if (s->sys->height == 720 && ((s->buf[1]>>2)&0x3) == 0) {
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mb_y -= (mb_y>17)?18:-72; /* shifting the Y coordinate down by 72/2 macro blocks */
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mb_y -= (mb_y>17)?18:-72; /* shifting the Y coordinate down by 72/2 macroblocks */
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}
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/* idct_put'ting luminance */
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@ -663,7 +663,7 @@ static av_always_inline void dv_set_class_number(DCTELEM* blk, EncBlockInfo* bi,
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method suggested in SMPTE 314M Table 22, and an improved
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method. The SMPTE method is very conservative; it assigns class
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3 (i.e. severe quantization) to any block where the largest AC
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component is greater than 36. ffmpeg's DV encoder tracks AC bit
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component is greater than 36. FFmpeg's DV encoder tracks AC bit
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consumption precisely, so there is no need to bias most blocks
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towards strongly lossy compression. Instead, we assign class 2
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to most blocks, and use class 3 only when strictly necessary
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@ -671,7 +671,7 @@ static av_always_inline void dv_set_class_number(DCTELEM* blk, EncBlockInfo* bi,
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#if 0 /* SMPTE spec method */
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static const int classes[] = {12, 24, 36, 0xffff};
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#else /* improved ffmpeg method */
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#else /* improved FFmpeg method */
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static const int classes[] = {-1, -1, 255, 0xffff};
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#endif
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int max=classes[0];
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@ -1176,7 +1176,7 @@ static void dv_format_frame(DVVideoContext* c, uint8_t* buf)
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buf += 77; /* audio control & shuffled PCM audio */
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}
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buf += dv_write_dif_id(dv_sect_video, chan, i, j, buf);
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buf += 77; /* 1 video macro block: 1 bytes control
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buf += 77; /* 1 video macroblock: 1 bytes control
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4 * 14 bytes Y 8x8 data
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10 bytes Cr 8x8 data
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10 bytes Cb 8x8 data */
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@ -48,13 +48,13 @@ typedef struct DVprofile {
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int height; /* picture height in pixels */
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int width; /* picture width in pixels */
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AVRational sar[2]; /* sample aspect ratios for 4:3 and 16:9 */
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const uint16_t *video_place; /* positions of all DV macro blocks */
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const uint16_t *video_place; /* positions of all DV macroblocks */
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enum PixelFormat pix_fmt; /* picture pixel format */
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int bpm; /* blocks per macroblock */
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const uint8_t *block_sizes; /* AC block sizes, in bits */
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int audio_stride; /* size of audio_shuffle table */
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int audio_min_samples[3];/* min ammount of audio samples */
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/* for 48Khz, 44.1Khz and 32Khz */
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int audio_min_samples[3];/* min amount of audio samples */
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/* for 48kHz, 44.1kHz and 32kHz */
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int audio_samples_dist[5];/* how many samples are supposed to be */
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/* in each frame in a 5 frames window */
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const uint8_t (*audio_shuffle)[9]; /* PCM shuffling table */
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@ -323,7 +323,7 @@ static const uint8_t dv100_qstep[16] = {
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2, 3, 4, 5, 6, 7, 8, 16, 18, 20, 22, 24, 28, 52
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};
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/* NOTE: I prefer hardcoding the positioning of dv blocks, it is
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/* NOTE: I prefer hardcoding the positioning of DV blocks, it is
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simpler :-) */
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static const uint16_t dv_place_420[1620] = {
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@ -6175,7 +6175,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 6,
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.block_sizes = block_sizes_dv2550,
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.audio_stride = 90,
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
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.audio_shuffle = dv_audio_shuffle525,
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},
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@ -6195,7 +6195,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 6,
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.block_sizes = block_sizes_dv2550,
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.audio_stride = 108,
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
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.audio_shuffle = dv_audio_shuffle625,
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},
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@ -6215,7 +6215,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 6,
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.block_sizes = block_sizes_dv2550,
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.audio_stride = 108,
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
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.audio_shuffle = dv_audio_shuffle625,
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},
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@ -6235,7 +6235,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 6,
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.block_sizes = block_sizes_dv2550,
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.audio_stride = 90,
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
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.audio_shuffle = dv_audio_shuffle525,
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},
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@ -6255,7 +6255,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 6,
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.block_sizes = block_sizes_dv2550,
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.audio_stride = 108,
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
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.audio_shuffle = dv_audio_shuffle625,
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},
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@ -6275,7 +6275,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 8,
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.block_sizes = block_sizes_dv100,
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.audio_stride = 90,
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
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.audio_shuffle = dv_audio_shuffle525,
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},
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@ -6295,7 +6295,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 8,
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.block_sizes = block_sizes_dv100,
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.audio_stride = 108,
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1896, 1742, 1264 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1920, 1920, 1920, 1920, 1920 },
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.audio_shuffle = dv_audio_shuffle625,
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},
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@ -6315,7 +6315,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 8,
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.block_sizes = block_sizes_dv100,
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.audio_stride = 90,
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
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.audio_shuffle = dv_audio_shuffle525,
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},
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@ -6335,7 +6335,7 @@ static const DVprofile dv_profiles[] = {
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.bpm = 8,
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.block_sizes = block_sizes_dv100,
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.audio_stride = 90,
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32Khz */
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.audio_min_samples = { 1580, 1452, 1053 }, /* for 48, 44.1 and 32kHz */
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.audio_samples_dist = { 1600, 1602, 1602, 1602, 1602 }, /* per SMPTE-314M */
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.audio_shuffle = dv_audio_shuffle525,
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}
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@ -33,7 +33,7 @@ VLC ff_msmp4_mb_i_vlc;
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VLC ff_msmp4_dc_luma_vlc[2];
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VLC ff_msmp4_dc_chroma_vlc[2];
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/* intra picture macro block coded block pattern */
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/* intra picture macroblock coded block pattern */
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const uint16_t ff_msmp4_mb_i_table[64][2] = {
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{ 0x1, 1 },{ 0x17, 6 },{ 0x9, 5 },{ 0x5, 5 },
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{ 0x6, 5 },{ 0x47, 9 },{ 0x20, 7 },{ 0x10, 7 },
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@ -53,7 +53,7 @@ const uint16_t ff_msmp4_mb_i_table[64][2] = {
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{ 0xd, 8 },{ 0x713, 13 },{ 0x1da, 10 },{ 0x169, 10 },
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};
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/* non intra picture macro block coded block pattern + mb type */
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/* non intra picture macroblock coded block pattern + mb type */
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const uint32_t table_mb_non_intra[128][2] = {
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{ 0x40, 7 },{ 0x13c9, 13 },{ 0x9fd, 12 },{ 0x1fc, 15 },
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{ 0x9fc, 12 },{ 0xa83, 18 },{ 0x12d34, 17 },{ 0x83bc, 16 },
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@ -304,7 +304,7 @@ static const int8_t table0_run[132] = {
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23, 24, 25, 26,
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};
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/* vlc table 1, for intra chroma and P macro blocks */
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/* vlc table 1, for intra chroma and P macroblocks */
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static const uint16_t table1_vlc[149][2] = {
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{ 0x4, 3 },{ 0x14, 5 },{ 0x17, 7 },{ 0x7f, 8 },
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@ -49,7 +49,7 @@ extern VLC ff_msmp4_mb_i_vlc;
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extern VLC ff_msmp4_dc_luma_vlc[2];
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extern VLC ff_msmp4_dc_chroma_vlc[2];
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/* intra picture macro block coded block pattern */
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/* intra picture macroblock coded block pattern */
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extern const uint16_t ff_msmp4_mb_i_table[64][2];
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extern const uint8_t cbpy_tab[16][2];
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@ -250,7 +250,7 @@ void rv10_encode_picture_header(MpegEncContext *s, int picture_number)
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/* specific MPEG like DC coding not used */
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}
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/* if multiple packets per frame are sent, the position at which
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to display the macro blocks is coded here */
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to display the macroblocks is coded here */
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if(!full_frame){
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put_bits(&s->pb, 6, 0); /* mb_x */
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put_bits(&s->pb, 6, 0); /* mb_y */
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@ -352,7 +352,7 @@ static int rv10_decode_picture_header(MpegEncContext *s)
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}
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}
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/* if multiple packets per frame are sent, the position at which
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to display the macro blocks is coded here */
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to display the macroblocks is coded here */
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mb_xy= s->mb_x + s->mb_y*s->mb_width;
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if(show_bits(&s->gb, 12)==0 || (mb_xy && mb_xy < s->mb_num)){
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@ -112,7 +112,7 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
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return 0;
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smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
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freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
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freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
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quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
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if (quant > 1)
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@ -145,8 +145,8 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
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if (of*2 >= size)
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continue;
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pcm[of*2] = frame[d+1]; // FIXME: may be we have to admit
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pcm[of*2+1] = frame[d]; // that DV is a big endian PCM
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pcm[of*2] = frame[d+1]; // FIXME: maybe we have to admit
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pcm[of*2+1] = frame[d]; // that DV is a big-endian PCM
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if (pcm[of*2+1] == 0x80 && pcm[of*2] == 0x00)
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pcm[of*2+1] = 0;
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} else { /* 12bit quantization */
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@ -161,12 +161,12 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
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if (of*2 >= size)
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continue;
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pcm[of*2] = lc & 0xff; // FIXME: may be we have to admit
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pcm[of*2+1] = lc >> 8; // that DV is a big endian PCM
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pcm[of*2] = lc & 0xff; // FIXME: maybe we have to admit
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pcm[of*2+1] = lc >> 8; // that DV is a big-endian PCM
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of = sys->audio_shuffle[i%half_ch+half_ch][j] +
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(d - 8)/3 * sys->audio_stride;
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pcm[of*2] = rc & 0xff; // FIXME: may be we have to admit
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pcm[of*2+1] = rc >> 8; // that DV is a big endian PCM
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pcm[of*2] = rc & 0xff; // FIXME: maybe we have to admit
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pcm[of*2+1] = rc >> 8; // that DV is a big-endian PCM
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++d;
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}
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}
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@ -196,7 +196,7 @@ static int dv_extract_audio_info(DVDemuxContext* c, uint8_t* frame)
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}
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smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
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freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
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freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48kHz, 1 - 44,1kHz, 2 - 32kHz */
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stype = (as_pack[3] & 0x1f); /* 0 - 2CH, 2 - 4CH, 3 - 8CH */
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quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
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@ -38,7 +38,7 @@ struct DVMuxContext {
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const DVprofile* sys; /* Current DV profile. E.g.: 525/60, 625/50 */
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int n_ast; /* Number of stereo audio streams (up to 2) */
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AVStream *ast[2]; /* Stereo audio streams */
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AVFifoBuffer audio_data[2]; /* Fifo for storing excessive amounts of PCM */
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AVFifoBuffer audio_data[2]; /* FIFO for storing excessive amounts of PCM */
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int frames; /* Number of a current frame */
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time_t start_time; /* Start time of recording */
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int has_audio; /* frame under contruction has audio */
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@ -117,7 +117,7 @@ static int dv_write_pack(enum dv_pack_type pack_id, DVMuxContext *c, uint8_t* bu
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(c->sys->n_difchan & 2); /* definition: 0 -- 25Mbps, 2 -- 50Mbps */
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buf[4] = (1 << 7) | /* emphasis: 1 -- off */
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(0 << 6) | /* emphasis time constant: 0 -- reserved */
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(0 << 3) | /* frequency: 0 -- 48Khz, 1 -- 44,1Khz, 2 -- 32Khz */
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(0 << 3) | /* frequency: 0 -- 48kHz, 1 -- 44,1kHz, 2 -- 32kHz */
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0; /* quantization: 0 -- 16bit linear, 1 -- 12bit nonlinear */
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va_end(ap);
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break;
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@ -189,8 +189,8 @@ static void dv_inject_audio(DVMuxContext *c, int channel, uint8_t* frame_ptr)
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if (of*2 >= size)
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continue;
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frame_ptr[d] = av_fifo_peek(&c->audio_data[channel], of*2+1); // FIXME: may be we have to admit
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frame_ptr[d+1] = av_fifo_peek(&c->audio_data[channel], of*2); // that DV is a big endian PCM
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frame_ptr[d] = av_fifo_peek(&c->audio_data[channel], of*2+1); // FIXME: maybe we have to admit
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frame_ptr[d+1] = av_fifo_peek(&c->audio_data[channel], of*2); // that DV is a big-endian PCM
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}
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frame_ptr += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
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}
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@ -365,7 +365,7 @@ static int dv_write_header(AVFormatContext *s)
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if (!dv_init_mux(s)) {
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av_log(s, AV_LOG_ERROR, "Can't initialize DV format!\n"
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"Make sure that you supply exactly two streams:\n"
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" video: 25fps or 29.97fps, audio: 2ch/48Khz/PCM\n"
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" video: 25fps or 29.97fps, audio: 2ch/48kHz/PCM\n"
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" (50Mbps allows an optional second audio stream)\n");
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return -1;
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}
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@ -1201,7 +1201,7 @@ static int mpegts_probe(AVProbeData *p)
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#endif
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}
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/* return the 90 kHz PCR and the extension for the 27 MHz PCR. return
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/* return the 90kHz PCR and the extension for the 27MHz PCR. return
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(-1) if not available */
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static int parse_pcr(int64_t *ppcr_high, int *ppcr_low,
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const uint8_t *packet)
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@ -104,7 +104,7 @@ void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
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memcpy(q, buf1, len);
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q += len;
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/* 90 KHz time stamp */
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/* 90kHz time stamp */
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s->timestamp = s->cur_timestamp;
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ff_rtp_send_data(s1, s->buf, q - s->buf, (len == size));
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