spelling cosmetics

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@15518 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

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

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 */

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,
     }

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 },

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];

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)){

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 */
 

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;
     }

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)

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));