Merge commit '511cf612ac979f536fd65e14603a87ca5ad435f3'

* commit '511cf612ac979f536fd65e14603a87ca5ad435f3':
  miscellaneous typo fixes

Conflicts:
	libavcodec/4xm.c
	libavcodec/lagarith.c
	libavcodec/parser.c
	libavcodec/ratecontrol.c
	libavcodec/shorten.c
	libavcodec/vda_h264.c
	libavformat/dvenc.c
	libavformat/wtv.c
	tools/patcheck

Merged-by: Michael Niedermayer <michaelni@gmx.at>

configure

@@ -1460,7 +1460,7 @@ 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

@@ -170,7 +170,7 @@ 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/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
+Finding the new best paths and scores for each point of our new column is
-trivial given we know the previous column best pathes and scores:
+trivial given we know the previous column best paths and scores:
 
     O-----0-----8
      \

libavcodec/4xm.c

@@ -842,7 +842,7 @@ 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\n");
+            av_log(f->avctx, AV_LOG_ERROR, "realloc failure\n");
             return -1;
         }
 

libavcodec/aacpsy.c

@@ -597,7 +597,7 @@ 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]);
@@ -617,7 +617,7 @@ 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

@@ -65,7 +65,7 @@ void ff_acelp_filter_init_mips(ACELPFContext *c);
  * 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/bitstream.c

@@ -172,7 +172,7 @@ 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/ffv1dec.c

@@ -757,7 +757,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame, AVPac
     } 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

@@ -644,7 +644,7 @@ 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/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

@@ -198,7 +198,7 @@ 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
+             *                          // since the compression change is negligible and fixing it
              *                          // breaks backwards compatibility
              *      b =- (signed int)b;
              *      b &= 0xFF;

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

@@ -341,7 +341,7 @@ 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/parser.c

@@ -95,7 +95,7 @@ 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 doesn't 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

@@ -372,7 +372,7 @@ 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

@@ -816,7 +816,7 @@ static int init_pass2(MpegEncContext *s)
     AVCodecContext *a= s->avctx;
     int i, toobig;
     double fps= get_fps(s->avctx);
-    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

@@ -406,7 +406,7 @@ 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

@@ -740,7 +740,7 @@ 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

@@ -526,7 +526,8 @@ 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 definition 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/vorbisdec.c

@@ -1235,7 +1235,7 @@ 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/wmaprodec.c

@@ -1099,7 +1099,7 @@ 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 */

libavformat/dvenc.c

@@ -51,9 +51,9 @@ 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_audio;     /* frame under construction has audio */
-    int               has_video;     /* frame under contruction has video */
+    int               has_video;     /* frame under construction has video */
-    uint8_t           frame_buf[DV_MAX_FRAME_SIZE]; /* frame under contruction */
+    uint8_t           frame_buf[DV_MAX_FRAME_SIZE]; /* frame under construction */
     AVTimecode        tc;            /* timecode context */
 };
 

libavformat/rtpdec_jpeg.c

@@ -370,7 +370,7 @@ static int jpeg_parse_packet(AVFormatContext *ctx, PayloadContext *jpeg,
         /* Prepare the JPEG packet. */
         if ((ret = ff_rtp_finalize_packet(pkt, &jpeg->frame, st->index)) < 0) {
             av_log(ctx, AV_LOG_ERROR,
-                   "Error occured when getting frame buffer.\n");
+                   "Error occurred when getting frame buffer.\n");
             return ret;
         }
 

libavformat/smoothstreamingenc.c

@@ -51,7 +51,7 @@ typedef struct {
     char dirname[1024];
     uint8_t iobuf[32768];
     URLContext *out;  // Current output stream where all output is written
-    URLContext *out2; // Auxillary output stream where all output also is written
+    URLContext *out2; // Auxiliary output stream where all output is also written
     URLContext *tail_out; // The actual main output stream, if we're currently seeked back to write elsewhere
     int64_t tail_pos, cur_pos, cur_start_pos;
     int packets_written;

libavformat/spdifenc.c

@@ -339,7 +339,7 @@ static int spdif_header_mpeg(AVFormatContext *s, AVPacket *pkt)
         ctx->data_type  = mpeg_data_type [version & 1][layer];
         ctx->pkt_offset = spdif_mpeg_pkt_offset[version & 1][layer];
     }
-    // TODO Data type dependant info (normal/karaoke, dynamic range control)
+    // TODO Data type dependent info (normal/karaoke, dynamic range control)
     return 0;
 }
 

libavresample/avresample-test.c

@@ -100,7 +100,7 @@ static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
         a += M_PI * 1000.0 * 2.0 / sample_rate;
     }
 
-    /* 1 second of varing frequency between 100 and 10000 Hz */
+    /* 1 second of varying frequency between 100 and 10000 Hz */
     a = 0;
     for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
         v = sin(a) * 0.30;

libswscale/ppc/yuv2yuv_altivec.c

@@ -1,5 +1,5 @@
 /*
- * AltiVec-enhanced yuv-to-yuv convertion routines.
+ * AltiVec-enhanced yuv-to-yuv conversion routines.
  *
  * Copyright (C) 2004 Romain Dolbeau <romain@dolbeau.org>
  * based on the equivalent C code in swscale.c

libswscale/swscale.c

@@ -148,7 +148,7 @@ static void hScale8To19_c(SwsContext *c, int16_t *_dst, int dstW,
     }
 }
 
-// FIXME all pal and rgb srcFormats could do this convertion as well
+// FIXME all pal and rgb srcFormats could do this conversion as well
 // FIXME all scalers more complex than bilinear could do half of this transform
 static void chrRangeToJpeg_c(int16_t *dstU, int16_t *dstV, int width)
 {

tests/audiogen.c

@@ -189,7 +189,7 @@ int main(int argc, char **argv)
         a += (1000 * FRAC_ONE) / sample_rate;
     }
 
-    /* 1 second of varing frequency between 100 and 10000 Hz */
+    /* 1 second of varying frequency between 100 and 10000 Hz */
     a = 0;
     for (i = 0; i < 1 * sample_rate; i++) {
         v = (int_cos(a) * 10000) >> FRAC_BITS;

tools/patcheck

@@ -158,7 +158,7 @@ cat $* | tr '\n' '@' | $EGREP --color=always -o '[^a-zA-Z0-9_]([a-zA-Z0-9_]*) *=
 cat $TMP | tr '@' '\n'
 
 
-# doesnt work
+# does not work
 #cat $* | tr '\n' '@' | $EGREP -o '[^a-zA-Z_0-9]([a-zA-Z][a-zA-Z_0-9]*) *=[^=].*\1' | $EGREP -o '[^a-zA-Z_0-9]([a-zA-Z][a-zA-Z_0-9]*) *=[^=].*\1 *=[^=]'  >$TMP && printf "\nPossibly written 2x before read\n"
 #cat $TMP | tr '@' '\n'