diff options
author | Diego Biurrun <diego@biurrun.de> | 2008-10-02 20:28:58 +0400 |
---|---|---|
committer | Diego Biurrun <diego@biurrun.de> | 2008-10-02 20:28:58 +0400 |
commit | 910f02a05434bb6d8b946284c0da254a44707a83 (patch) | |
tree | 5a406a3892ace5cab3e071b32b6cad708577bd2f | |
parent | fb65d2ca84d79fb1c5a5708555c23e1d289b5c92 (diff) |
spelling cosmetics
Originally committed as revision 15518 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r-- | doc/ffmpeg-doc.texi | 2 | ||||
-rw-r--r-- | libavcodec/dv.c | 10 | ||||
-rw-r--r-- | libavcodec/dvdata.h | 26 | ||||
-rw-r--r-- | libavcodec/msmpeg4data.c | 6 | ||||
-rw-r--r-- | libavcodec/msmpeg4data.h | 2 | ||||
-rw-r--r-- | libavcodec/rv10.c | 4 | ||||
-rw-r--r-- | libavformat/dv.c | 16 | ||||
-rw-r--r-- | libavformat/dvenc.c | 10 | ||||
-rw-r--r-- | libavformat/mpegts.c | 2 | ||||
-rw-r--r-- | libavformat/rtp_mpv.c | 2 |
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)); |