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ffmpeg.wasm-core
Commit 910f02a0 authored by Diego Biurrun's avatar Diego Biurrun
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spelling cosmetics 

Originally committed as revision 15518 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent fb65d2ca
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doc/ffmpeg-doc.texi
View file @ 910f02a0
......@@ -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
......
libavcodec/dv.c
View file @ 910f02a0
......@@ -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;
......@@ -493,7 +493,7 @@ static inline void dv_decode_video_segment(DVVideoContext *s,
mb_y = v >> 8;
/* We work with 720p frames split in half. The odd half-frame (chan==2,3) is displaced :-( */
if (s->sys->height == 720 && ((s->buf[1]>>2)&0x3) == 0) {
mb_y -= (mb_y>17)?18:-72; /* shifting the Y coordinate down by 72/2 macro blocks */
mb_y -= (mb_y>17)?18:-72; /* shifting the Y coordinate down by 72/2 macroblocks */
}
/* idct_put'ting luminance */
......@@ -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];
......@@ -1176,7 +1176,7 @@ static void dv_format_frame(DVVideoContext* c, uint8_t* buf)
buf += 77; /* audio control & shuffled PCM audio */
}
buf += dv_write_dif_id(dv_sect_video, chan, i, j, buf);
buf += 77; /* 1 video macro block: 1 bytes control
buf += 77; /* 1 video macroblock: 1 bytes control
4 * 14 bytes Y 8x8 data
10 bytes Cr 8x8 data
10 bytes Cb 8x8 data */
......
libavcodec/dvdata.h
View file @ 910f02a0
......@@ -48,13 +48,13 @@ typedef struct DVprofile {
int height; /* picture height in pixels */
int width; /* picture width in pixels */
AVRational sar[2]; /* sample aspect ratios for 4:3 and 16:9 */
const uint16_t *video_place; /* positions of all DV macro blocks */
const uint16_t *video_place; /* positions of all DV macroblocks */
enum PixelFormat pix_fmt; /* picture pixel format */
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,
}
......
libavcodec/msmpeg4data.c
View file @ 910f02a0
......@@ -33,7 +33,7 @@ VLC ff_msmp4_mb_i_vlc;
VLC ff_msmp4_dc_luma_vlc[2];
VLC ff_msmp4_dc_chroma_vlc[2];
/* intra picture macro block coded block pattern */
/* intra picture macroblock coded block pattern */
const uint16_t ff_msmp4_mb_i_table[64][2] = {
{ 0x1, 1 },{ 0x17, 6 },{ 0x9, 5 },{ 0x5, 5 },
{ 0x6, 5 },{ 0x47, 9 },{ 0x20, 7 },{ 0x10, 7 },
......@@ -53,7 +53,7 @@ const uint16_t ff_msmp4_mb_i_table[64][2] = {
{ 0xd, 8 },{ 0x713, 13 },{ 0x1da, 10 },{ 0x169, 10 },
};
/* non intra picture macro block coded block pattern + mb type */
/* non intra picture macroblock coded block pattern + mb type */
const uint32_t table_mb_non_intra[128][2] = {
{ 0x40, 7 },{ 0x13c9, 13 },{ 0x9fd, 12 },{ 0x1fc, 15 },
{ 0x9fc, 12 },{ 0xa83, 18 },{ 0x12d34, 17 },{ 0x83bc, 16 },
......@@ -304,7 +304,7 @@ static const int8_t table0_run[132] = {
23, 24, 25, 26,
};
/* vlc table 1, for intra chroma and P macro blocks */
/* vlc table 1, for intra chroma and P macroblocks */
static const uint16_t table1_vlc[149][2] = {
{ 0x4, 3 },{ 0x14, 5 },{ 0x17, 7 },{ 0x7f, 8 },
......
libavcodec/msmpeg4data.h
View file @ 910f02a0
......@@ -49,7 +49,7 @@ extern VLC ff_msmp4_mb_i_vlc;
extern VLC ff_msmp4_dc_luma_vlc[2];
extern VLC ff_msmp4_dc_chroma_vlc[2];
/* intra picture macro block coded block pattern */
/* intra picture macroblock coded block pattern */
extern const uint16_t ff_msmp4_mb_i_table[64][2];
extern const uint8_t cbpy_tab[16][2];
......
libavcodec/rv10.c
View file @ 910f02a0
......@@ -250,7 +250,7 @@ void rv10_encode_picture_header(MpegEncContext *s, int picture_number)
/* specific MPEG like DC coding not used */
}
/* if multiple packets per frame are sent, the position at which
to display the macro blocks is coded here */
to display the macroblocks is coded here */
if(!full_frame){
put_bits(&s->pb, 6, 0); /* mb_x */
put_bits(&s->pb, 6, 0); /* mb_y */
......@@ -352,7 +352,7 @@ static int rv10_decode_picture_header(MpegEncContext *s)
}
}
/* if multiple packets per frame are sent, the position at which
to display the macro blocks is coded here */
to display the macroblocks is coded here */
mb_xy= s->mb_x + s->mb_y*s->mb_width;
if(show_bits(&s->gb, 12)==0 || (mb_xy && mb_xy < s->mb_num)){
......
libavformat/dv.c
View file @ 910f02a0
......@@ -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)
......@@ -145,8 +145,8 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
if (of*2 >= size)
continue;
pcm[of*2] = frame[d+1]; // FIXME: may be we have to admit
pcm[of*2+1] = frame[d]; // that DV is a big endian PCM
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
if (pcm[of*2+1] == 0x80 && pcm[of*2] == 0x00)
pcm[of*2+1] = 0;
} else { /* 12bit quantization */
......@@ -161,12 +161,12 @@ static int dv_extract_audio(uint8_t* frame, uint8_t* ppcm[4],
if (of*2 >= size)
continue;
pcm[of*2] = lc & 0xff; // FIXME: may be we have to admit
pcm[of*2+1] = lc >> 8; // that DV is a big endian PCM
pcm[of*2] = lc & 0xff; // FIXME: maybe we have to admit
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: may be we have to admit
pcm[of*2+1] = rc >> 8; // that DV is a big endian PCM
pcm[of*2] = rc & 0xff; // FIXME: maybe we have to admit
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 */
......
libavformat/dvenc.c
View file @ 910f02a0
......@@ -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;
......@@ -189,8 +189,8 @@ static void dv_inject_audio(DVMuxContext *c, int channel, uint8_t* frame_ptr)
if (of*2 >= size)
continue;
frame_ptr[d] = av_fifo_peek(&c->audio_data[channel], of*2+1); // FIXME: may be 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] = 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 += 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;
}
......
libavformat/mpegts.c
View file @ 910f02a0
......@@ -1201,7 +1201,7 @@ static int mpegts_probe(AVProbeData *p)
#endif
}
/* return the 90 kHz PCR and the extension for the 27 MHz PCR. return
/* return the 90kHz PCR and the extension for the 27MHz PCR. return
(-1) if not available */
static int parse_pcr(int64_t *ppcr_high, int *ppcr_low,
const uint8_t *packet)
......
libavformat/rtp_mpv.c
View file @ 910f02a0
......@@ -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));
......
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