miscellaneous typo fixes

configure

@@ -1305,7 +1305,7 @@ HAVE_LIST="
     xmm_clobbers
 "
 
-# options emitted with CONFIG_ prefix but not available on command line
+# options emitted with CONFIG_ prefix but not available on the command line
 CONFIG_EXTRA="
     aandcttables
     ac3dsp

doc/Doxyfile

@@ -288,7 +288,7 @@ TYPEDEF_HIDES_STRUCT   = NO
 # causing a significant performance penality.
 # If the system has enough physical memory increasing the cache will improve the
 # performance by keeping more symbols in memory. Note that the value works on
-# a logarithmic scale so increasing the size by one will rougly double the
+# a logarithmic scale so increasing the size by one will roughly double the
 # memory usage. The cache size is given by this formula:
 # 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
 # corresponding to a cache size of 2^16 = 65536 symbols

doc/developer.texi

@@ -201,7 +201,7 @@ For exported names, each library has its own prefixes. Just check the existing
 code and name accordingly.
 @end itemize
 
-@subsection Miscellanous conventions
+@subsection Miscellaneous conventions
 @itemize @bullet
 @item
 fprintf and printf are forbidden in libavformat and libavcodec,

doc/indevs.texi

@@ -300,7 +300,7 @@ The filename passed as input has the syntax:
 
 @var{hostname}:@var{display_number}.@var{screen_number} specifies the
 X11 display name of the screen to grab from. @var{hostname} can be
-ommitted, and defaults to "localhost". The environment variable
+omitted, and defaults to "localhost". The environment variable
 @env{DISPLAY} contains the default display name.
 
 @var{x_offset} and @var{y_offset} specify the offsets of the grabbed

doc/rate_distortion.txt

@@ -23,7 +23,7 @@ Let's consider the problem of minimizing:
 
 rate is the filesize
 distortion is the quality
-lambda is a fixed value choosen as a tradeoff between quality and filesize
+lambda is a fixed value chosen as a tradeoff between quality and filesize
 Is this equivalent to finding the best quality for a given max
 filesize? The answer is yes. For each filesize limit there is some lambda
 factor for which minimizing above will get you the best quality (using your

doc/viterbi.txt

@@ -85,8 +85,8 @@ here are some edges we could choose from:
      /        \
     O-----2--4--O
 
-Finding the new best pathes and scores for each point of our new column is
-trivial given we know the previous column best pathes and scores:
+Finding the new best paths and scores for each point of our new column is
+trivial given we know the previous column best paths and scores:
 
     O-----0-----8
      \

libavcodec/4xm.c

@@ -796,7 +796,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
                                      cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
         // explicit check needed as memcpy below might not catch a NULL
         if (!cfrm->data) {
-            av_log(f->avctx, AV_LOG_ERROR, "realloc falure");
+            av_log(f->avctx, AV_LOG_ERROR, "realloc failure");
             return -1;
         }
 

libavcodec/aacpsy.c

@@ -592,7 +592,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
     for (w = 0; w < wi->num_windows*16; w += 16) {
         AacPsyBand *bands = &pch->band[w];
 
-        //5.4.2.3 "Spreading" & 5.4.3 "Spreaded Energy Calculation"
+        /* 5.4.2.3 "Spreading" & 5.4.3 "Spread Energy Calculation" */
         spread_en[0] = bands[0].energy;
         for (g = 1; g < num_bands; g++) {
             bands[g].thr   = FFMAX(bands[g].thr,    bands[g-1].thr * coeffs[g].spread_hi[0]);
@@ -612,7 +612,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
                 band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
                                   PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
 
-            /* 5.6.1.3.1 "Prepatory steps of the perceptual entropy calculation" */
+            /* 5.6.1.3.1 "Preparatory steps of the perceptual entropy calculation" */
             pe += calc_pe_3gpp(band);
             a  += band->pe_const;
             active_lines += band->active_lines;

libavcodec/ac3dec.c

@@ -546,7 +546,7 @@ static void decode_transform_coeffs(AC3DecodeContext *s, int blk)
     for (ch = 1; ch <= s->channels; ch++) {
         /* transform coefficients for full-bandwidth channel */
         decode_transform_coeffs_ch(s, blk, ch, &m);
-        /* tranform coefficients for coupling channel come right after the
+        /* transform coefficients for coupling channel come right after the
            coefficients for the first coupled channel*/
         if (s->channel_in_cpl[ch])  {
             if (!got_cplchan) {

libavcodec/ac3enc.c

@@ -659,7 +659,7 @@ static void count_frame_bits_fixed(AC3EncodeContext *s)
      *   bit allocation parameters do not change between blocks
      *   no delta bit allocation
      *   no skipped data
-     *   no auxilliary data
+     *   no auxiliary data
      *   no E-AC-3 metadata
      */
 

libavcodec/acelp_filters.h

@@ -32,7 +32,7 @@
  * the coefficients are scaled by 2^15.
  * This array only contains the right half of the filter.
  * This filter is likely identical to the one used in G.729, though this
- * could not be determined from the original comments with certainity.
+ * could not be determined from the original comments with certainty.
  */
 extern const int16_t ff_acelp_interp_filter[61];
 

libavcodec/avcodec.h

@@ -2292,7 +2292,7 @@ typedef struct AVCodecContext {
 
     /**
      * ratecontrol qmin qmax limiting method
-     * 0-> clipping, 1-> use a nice continous function to limit qscale wthin qmin/qmax.
+     * 0-> clipping, 1-> use a nice continuous function to limit qscale wthin qmin/qmax.
      * - encoding: Set by user.
      * - decoding: unused
      */

libavcodec/bitstream.c

@@ -169,7 +169,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
         table[i][0] = -1; //codes
     }
 
-    /* first pass: map codes and compute auxillary table sizes */
+    /* first pass: map codes and compute auxiliary table sizes */
     for (i = 0; i < nb_codes; i++) {
         n = codes[i].bits;
         code = codes[i].code;

libavcodec/eac3dec.c

@@ -491,7 +491,7 @@ int ff_eac3_parse_header(AC3DecodeContext *s)
     s->skip_syntax       = get_bits1(gbc);
     parse_spx_atten_data = get_bits1(gbc);
 
-    /* coupling strategy occurance and coupling use per block */
+    /* coupling strategy occurrence and coupling use per block */
     num_cpl_blocks = 0;
     if (s->channel_mode > 1) {
         for (blk = 0; blk < s->num_blocks; blk++) {

libavcodec/ffv1dec.c

@@ -824,7 +824,7 @@ static int ffv1_decode_frame(AVCodecContext *avctx, void *data,
     } else {
         if (!f->key_frame_ok) {
             av_log(avctx, AV_LOG_ERROR,
-                   "Cant decode non keyframe without valid keyframe\n");
+                   "Cannot decode non-keyframe without valid keyframe\n");
             return AVERROR_INVALIDDATA;
         }
         p->key_frame = 0;

libavcodec/flicvideo.c

@@ -581,7 +581,7 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
                 }
 
                 /* Now FLX is strange, in that it is "byte" as opposed to "pixel" run length compressed.
-                 * This does not give us any good oportunity to perform word endian conversion
+                 * This does not give us any good opportunity to perform word endian conversion
                  * during decompression. So if it is required (i.e., this is not a LE target, we do
                  * a second pass over the line here, swapping the bytes.
                  */

libavcodec/g726.c

@@ -34,7 +34,7 @@
 /**
  * G.726 11bit float.
  * G.726 Standard uses rather odd 11bit floating point arithmentic for
- * numerous occasions. It's a mistery to me why they did it this way
+ * numerous occasions. It's a mystery to me why they did it this way
  * instead of simply using 32bit integer arithmetic.
  */
 typedef struct Float11 {

libavcodec/h264_direct.c

@@ -86,7 +86,7 @@ static void fill_colmap(H264Context *h, int map[2][16+32], int list, int field,
 
             if     (!interl)
                 poc |= 3;
-            else if( interl && (poc&3) == 3) //FIXME store all MBAFF references so this isnt needed
+            else if( interl && (poc&3) == 3) // FIXME: store all MBAFF references so this is not needed
                 poc= (poc&~3) + rfield + 1;
 
             for(j=start; j<end; j++){

libavcodec/indeo3data.h

@@ -235,7 +235,7 @@
 
 /**
  * Pack two delta values (a,b) into one 16bit word
- * according with endianess of the host machine.
+ * according with endianness of the host machine.
  */
 #if HAVE_BIGENDIAN
 #define PD(a,b) (((a) << 8) + (b))
@@ -282,7 +282,7 @@ static const int16_t delta_tab_3_5[79]  = { TAB_3_5 };
 
 /**
  * Pack four delta values (a,a,b,b) into one 32bit word
- * according with endianess of the host machine.
+ * according with endianness of the host machine.
  */
 #if HAVE_BIGENDIAN
 #define PD(a,b) (((a) << 24) + ((a) << 16) + ((b) << 8) + (b))

libavcodec/lagarith.c

@@ -198,8 +198,8 @@ static int lag_read_prob_header(lag_rac *rac, GetBitContext *gb)
             }
             /* Comment from reference source:
              * if (b & 0x80 == 0) {     // order of operations is 'wrong'; it has been left this way
-             *                          // since the compression change is negligable and fixing it
-             *                          // breaks backwards compatibilty
+             *                          // since the compression change is negligible and fixing it
+             *                          // breaks backwards compatibility
              *      b =- (signed int)b;
              *      b &= 0xFF;
              * } else {

libavcodec/libfdk-aacenc.c

@@ -257,7 +257,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
         }
         if ((err = aacEncoder_SetParam(s->handle, AACENC_BANDWIDTH,
                                        avctx->cutoff)) != AACENC_OK) {
-            av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwith to %d: %s\n",
+            av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwidth to %d: %s\n",
                    avctx->cutoff, aac_get_error(err));
             goto error;
         }

libavcodec/libtheoraenc.c

@@ -338,7 +338,7 @@ static int encode_frame(AVCodecContext* avc_context, AVPacket *pkt,
     memcpy(pkt->data, o_packet.packet, o_packet.bytes);
 
     // HACK: assumes no encoder delay, this is true until libtheora becomes
-    // multithreaded (which will be disabled unless explictly requested)
+    // multithreaded (which will be disabled unless explicitly requested)
     pkt->pts = pkt->dts = frame->pts;
     avc_context->coded_frame->key_frame = !(o_packet.granulepos & h->keyframe_mask);
     if (avc_context->coded_frame->key_frame)

libavcodec/mpeg4videoenc.c

@@ -89,7 +89,7 @@ static inline int get_block_rate(MpegEncContext * s, DCTELEM block[64], int bloc
  * @param[in,out] block MB coefficients, these will be restored
  * @param[in] dir ac prediction direction for each 8x8 block
  * @param[out] st scantable for each 8x8 block
- * @param[in] zigzag_last_index index refering to the last non zero coefficient in zigzag order
+ * @param[in] zigzag_last_index index referring to the last non zero coefficient in zigzag order
  */
 static inline void restore_ac_coeffs(MpegEncContext * s, DCTELEM block[6][64], const int dir[6], uint8_t *st[6], const int zigzag_last_index[6])
 {
@@ -120,7 +120,7 @@ static inline void restore_ac_coeffs(MpegEncContext * s, DCTELEM block[6][64], c
  * @param[in,out] block MB coefficients, these will be updated if 1 is returned
  * @param[in] dir ac prediction direction for each 8x8 block
  * @param[out] st scantable for each 8x8 block
- * @param[out] zigzag_last_index index refering to the last non zero coefficient in zigzag order
+ * @param[out] zigzag_last_index index referring to the last non zero coefficient in zigzag order
  */
 static inline int decide_ac_pred(MpegEncContext * s, DCTELEM block[6][64], const int dir[6], uint8_t *st[6], int zigzag_last_index[6])
 {

libavcodec/parser.c

@@ -96,7 +96,7 @@ void ff_fetch_timestamp(AVCodecParserContext *s, int off, int remove){
         if (   s->cur_offset + off >= s->cur_frame_offset[i]
             && (s->frame_offset < s->cur_frame_offset[i] ||
               (!s->frame_offset && !s->next_frame_offset)) // first field/frame
-            //check is disabled  because mpeg-ts doesnt send complete PES packets
+            // check disabled since MPEG-TS does not send complete PES packets
             && /*s->next_frame_offset + off <*/  s->cur_frame_end[i]){
             s->dts= s->cur_frame_dts[i];
             s->pts= s->cur_frame_pts[i];

libavcodec/pngenc.c

@@ -367,7 +367,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *pkt,
         int pass;
 
         for(pass = 0; pass < NB_PASSES; pass++) {
-            /* NOTE: a pass is completely omited if no pixels would be
+            /* NOTE: a pass is completely omitted if no pixels would be
                output */
             pass_row_size = ff_png_pass_row_size(pass, bits_per_pixel, avctx->width);
             if (pass_row_size > 0) {

libavcodec/ratecontrol.c

@@ -799,7 +799,7 @@ static int init_pass2(MpegEncContext *s)
     AVCodecContext *a= s->avctx;
     int i, toobig;
     double fps= 1/av_q2d(s->avctx->time_base);
-    double complexity[5]={0,0,0,0,0};   // aproximate bits at quant=1
+    double complexity[5]={0,0,0,0,0};   // approximate bits at quant=1
     uint64_t const_bits[5]={0,0,0,0,0}; // quantizer independent bits
     uint64_t all_const_bits;
     uint64_t all_available_bits= (uint64_t)(s->bit_rate*(double)rcc->num_entries/fps);

libavcodec/resample.c

@@ -350,7 +350,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
                              ibuf, istride, nb_samples1 * s->output_channels) < 0) {
             av_log(s->resample_context, AV_LOG_ERROR,
-                   "Audio sample format convertion failed\n");
+                   "Audio sample format conversion failed\n");
             return 0;
         }
     }

libavcodec/rv10.c

@@ -706,7 +706,7 @@ static int rv10_decode_frame(AVCodecContext *avctx,
             *got_frame = 1;
             ff_print_debug_info(s, pict);
         }
-        s->current_picture_ptr= NULL; //so we can detect if frame_end wasnt called (find some nicer solution...)
+        s->current_picture_ptr= NULL; // so we can detect if frame_end was not called (find some nicer solution...)
     }
 
     return avpkt->size;

libavcodec/shorten.c

@@ -528,7 +528,8 @@ static int shorten_decode_frame(AVCodecContext *avctx, void *data,
             /* get Rice code for residual decoding */
             if (cmd != FN_ZERO) {
                 residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE);
-                /* this is a hack as version 0 differed in defintion of get_sr_golomb_shorten */
+                /* This is a hack as version 0 differed in the definition
+                 * of get_sr_golomb_shorten(). */
                 if (s->version == 0)
                     residual_size--;
             }

libavcodec/thread.h

@@ -43,7 +43,7 @@ void ff_thread_flush(AVCodecContext *avctx);
  * Returns the next available frame in picture. *got_picture_ptr
  * will be 0 if none is available.
  * The return value on success is the size of the consumed packet for
- * compatiblity with avcodec_decode_video2(). This means the decoder
+ * compatibility with avcodec_decode_video2(). This means the decoder
  * has to consume the full packet.
  *
  * Parameters are the same as avcodec_decode_video2().

libavcodec/vda_h264.c

@@ -281,7 +281,7 @@ int ff_vda_create_decoder(struct vda_context *vda_ctx,
 #endif
 
     /* Each VCL NAL in the bistream sent to the decoder
-     * is preceeded by a 4 bytes length header.
+     * is preceded by a 4 bytes length header.
      * Change the avcC atom header if needed, to signal headers of 4 bytes. */
     if (extradata_size >= 4 && (extradata[4] & 0x03) != 0x03) {
         uint8_t *rw_extradata;

libavcodec/vorbisdec.c

@@ -1233,7 +1233,7 @@ static int vorbis_floor1_decode(vorbis_context *vc,
         if (highroom < lowroom) {
             room = highroom * 2;
         } else {
-            room = lowroom * 2;   // SPEC mispelling
+            room = lowroom * 2;   // SPEC misspelling
         }
         if (val) {
             floor1_flag[low_neigh_offs]  = 1;

libavcodec/vp8dsp.h

@@ -73,7 +73,7 @@ typedef struct VP8DSPContext {
      * second dimension: 0 if no vertical interpolation is needed;
      *                   1 4-tap vertical interpolation filter (my & 1)
      *                   2 6-tap vertical interpolation filter (!(my & 1))
-     * third dimension: same as second dimention, for horizontal interpolation
+     * third dimension: same as second dimension, for horizontal interpolation
      * so something like put_vp8_epel_pixels_tab[width>>3][2*!!my-(my&1)][2*!!mx-(mx&1)](..., mx, my)
      */
     vp8_mc_func put_vp8_epel_pixels_tab[3][3][3];

libavcodec/wmaprodec.c

@@ -533,7 +533,7 @@ static int decode_tilehdr(WMAProDecodeCtx *s)
     int c;
 
     /* Should never consume more than 3073 bits (256 iterations for the
-     * while loop when always the minimum amount of 128 samples is substracted
+     * while loop when always the minimum amount of 128 samples is subtracted
      * from missing samples in the 8 channel case).
      * 1 + BLOCK_MAX_SIZE * MAX_CHANNELS / BLOCK_MIN_SIZE * (MAX_CHANNELS  + 4)
      */
@@ -1089,7 +1089,7 @@ static int decode_subframe(WMAProDecodeCtx *s)
     s->channels_for_cur_subframe = 0;
     for (i = 0; i < s->avctx->channels; i++) {
         const int cur_subframe = s->channel[i].cur_subframe;
-        /** substract already processed samples */
+        /** subtract already processed samples */
         total_samples -= s->channel[i].decoded_samples;
 
         /** and count if there are multiple subframes that match our profile */

libavdevice/dv1394.h

@@ -186,7 +186,7 @@
    where copy_DV_frame() reads or writes on the dv1394 file descriptor
    (read/write mode) or copies data to/from the mmap ringbuffer and
    then calls ioctl(DV1394_SUBMIT_FRAMES) to notify dv1394 that new
-   frames are availble (mmap mode).
+   frames are available (mmap mode).
 
    reset_dv1394() is called in the event of a buffer
    underflow/overflow or a halt in the DV stream (e.g. due to a 1394

libavformat/avformat.h

@@ -1532,7 +1532,7 @@ enum AVCodecID av_guess_codec(AVOutputFormat *fmt, const char *short_name,
  * @ingroup libavf
  * @{
  *
- * Miscelaneous utility functions related to both muxing and demuxing
+ * Miscellaneous utility functions related to both muxing and demuxing
  * (or neither).
  */
 

libavformat/aviobuf.c

@@ -368,7 +368,7 @@ static void fill_buffer(AVIOContext *s)
     int max_buffer_size = s->max_packet_size ?
                           s->max_packet_size : IO_BUFFER_SIZE;
 
-    /* can't fill the buffer without read_packet, just set EOF if appropiate */
+    /* can't fill the buffer without read_packet, just set EOF if appropriate */
     if (!s->read_packet && s->buf_ptr >= s->buf_end)
         s->eof_reached = 1;
 

libavformat/dvenc.c

@@ -47,9 +47,9 @@ struct DVMuxContext {
     AVFifoBuffer     *audio_data[2]; /* FIFO for storing excessive amounts of PCM */
     int               frames;        /* current frame number */
     int64_t           start_time;    /* recording start time */
-    int               has_audio;     /* frame under contruction has audio */
-    int               has_video;     /* frame under contruction has video */
-    uint8_t           frame_buf[DV_MAX_FRAME_SIZE]; /* frame under contruction */
+    int               has_audio;     /* frame under construction has audio */
+    int               has_video;     /* frame under construction has video */
+    uint8_t           frame_buf[DV_MAX_FRAME_SIZE]; /* frame under construction */
 };
 
 static const int dv_aaux_packs_dist[12][9] = {

libavformat/hls.c

@@ -42,7 +42,7 @@
  * An apple http stream consists of a playlist with media segment files,
  * played sequentially. There may be several playlists with the same
  * video content, in different bandwidth variants, that are played in
- * parallel (preferrably only one bandwidth variant at a time). In this case,
+ * parallel (preferably only one bandwidth variant at a time). In this case,
  * the user supplied the url to a main playlist that only lists the variant
  * playlists.
  *