@@ -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

@@ -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 */

@@ -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,

     }

@@ -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 },

@@ -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];

@@ -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)){

@@ -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 */

@@ -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;

     }

@@ -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)

@@ -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));