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-rw-r--r--doc/ffmpeg-doc.texi2
-rw-r--r--libavcodec/dv.c10
-rw-r--r--libavcodec/dvdata.h26
-rw-r--r--libavcodec/msmpeg4data.c6
-rw-r--r--libavcodec/msmpeg4data.h2
-rw-r--r--libavcodec/rv10.c4
-rw-r--r--libavformat/dv.c16
-rw-r--r--libavformat/dvenc.c10
-rw-r--r--libavformat/mpegts.c2
-rw-r--r--libavformat/rtp_mpv.c2
10 files changed, 40 insertions, 40 deletions
diff --git a/doc/ffmpeg-doc.texi b/doc/ffmpeg-doc.texi
index 35e750d57b..69c265b51f 100644
--- a/doc/ffmpeg-doc.texi
+++ b/doc/ffmpeg-doc.texi
@@ -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
diff --git a/libavcodec/dv.c b/libavcodec/dv.c
index 876e93fe64..1067d73ee4 100644
--- a/libavcodec/dv.c
+++ b/libavcodec/dv.c
@@ -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 */
diff --git a/libavcodec/dvdata.h b/libavcodec/dvdata.h
index 6197b285a6..40206455f6 100644
--- a/libavcodec/dvdata.h
+++ b/libavcodec/dvdata.h
@@ -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,
}
diff --git a/libavcodec/msmpeg4data.c b/libavcodec/msmpeg4data.c
index 2b6e670c3b..da899b544b 100644
--- a/libavcodec/msmpeg4data.c
+++ b/libavcodec/msmpeg4data.c
@@ -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 },
diff --git a/libavcodec/msmpeg4data.h b/libavcodec/msmpeg4data.h
index 5dc8846d70..45c589c268 100644
--- a/libavcodec/msmpeg4data.h
+++ b/libavcodec/msmpeg4data.h
@@ -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];
diff --git a/libavcodec/rv10.c b/libavcodec/rv10.c
index 938e44694c..0c05147229 100644
--- a/libavcodec/rv10.c
+++ b/libavcodec/rv10.c
@@ -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)){
diff --git a/libavformat/dv.c b/libavformat/dv.c
index 815caf866d..22d359b031 100644
--- a/libavformat/dv.c
+++ b/libavformat/dv.c
@@ -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 */
diff --git a/libavformat/dvenc.c b/libavformat/dvenc.c
index 7edd7a842b..468fed3112 100644
--- a/libavformat/dvenc.c
+++ b/libavformat/dvenc.c
@@ -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;
}
diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c
index 9f41d396a9..48aca43d46 100644
--- a/libavformat/mpegts.c
+++ b/libavformat/mpegts.c
@@ -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)
diff --git a/libavformat/rtp_mpv.c b/libavformat/rtp_mpv.c
index 2c67f058a0..59d8c181a2 100644
--- a/libavformat/rtp_mpv.c
+++ b/libavformat/rtp_mpv.c
@@ -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));
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-01 22:29:27 +0000