diff options
35 files changed, 41 insertions, 40 deletions
@@ -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 diff --git a/doc/Doxyfile b/doc/Doxyfile index 9e12ab04af..7e6d0f56fd 100644 --- a/doc/Doxyfile +++ b/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 diff --git a/doc/developer.texi b/doc/developer.texi index b0e5216ed4..e75f3b9403 100644 --- a/doc/developer.texi +++ b/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, diff --git a/doc/rate_distortion.txt b/doc/rate_distortion.txt index a7d2c878b2..e9711c2d5c 100644 --- a/doc/rate_distortion.txt +++ b/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 diff --git a/doc/viterbi.txt b/doc/viterbi.txt index 5362a0b765..97825462cc 100644 --- a/doc/viterbi.txt +++ b/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 \ diff --git a/libavcodec/4xm.c b/libavcodec/4xm.c index 8fa214f1b4..de5c9f53ee 100644 --- a/libavcodec/4xm.c +++ b/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; } diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c index fa562b34b6..d77b3de4e4 100644 --- a/libavcodec/aacpsy.c +++ b/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; diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 4ca735f536..ea4a21809f 100644 --- a/libavcodec/ac3dec.c +++ b/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) { diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c index 22b6857847..de8defceb8 100644 --- a/libavcodec/ac3enc.c +++ b/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 */ diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h index 56197bcc18..7a3061bd1f 100644 --- a/libavcodec/acelp_filters.h +++ b/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]; diff --git a/libavcodec/bitstream.c b/libavcodec/bitstream.c index bf131e9a0b..6bcdadb9c4 100644 --- a/libavcodec/bitstream.c +++ b/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; diff --git a/libavcodec/ffv1dec.c b/libavcodec/ffv1dec.c index 2d03085873..a1da544697 100644 --- a/libavcodec/ffv1dec.c +++ b/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; diff --git a/libavcodec/flicvideo.c b/libavcodec/flicvideo.c index 2a117d6da6..eb70249ef7 100644 --- a/libavcodec/flicvideo.c +++ b/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. */ diff --git a/libavcodec/h264_direct.c b/libavcodec/h264_direct.c index 99c2ec1b97..1f2081017b 100644 --- a/libavcodec/h264_direct.c +++ b/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++){ diff --git a/libavcodec/indeo3data.h b/libavcodec/indeo3data.h index 0b5648eb20..e7e28a3b45 100644 --- a/libavcodec/indeo3data.h +++ b/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)) diff --git a/libavcodec/lagarith.c b/libavcodec/lagarith.c index 3a1791785f..486e326a0f 100644 --- a/libavcodec/lagarith.c +++ 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; diff --git a/libavcodec/libfdk-aacenc.c b/libavcodec/libfdk-aacenc.c index c2d8a2be25..196fcb5a4d 100644 --- a/libavcodec/libfdk-aacenc.c +++ b/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; } diff --git a/libavcodec/libtheoraenc.c b/libavcodec/libtheoraenc.c index 5a866740c8..14197236b1 100644 --- a/libavcodec/libtheoraenc.c +++ b/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) diff --git a/libavcodec/parser.c b/libavcodec/parser.c index 3b4715035a..f7cb5cfa67 100644 --- a/libavcodec/parser.c +++ b/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]; diff --git a/libavcodec/pngenc.c b/libavcodec/pngenc.c index c91f28941f..7ba14b2d0d 100644 --- a/libavcodec/pngenc.c +++ b/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) { diff --git a/libavcodec/ratecontrol.c b/libavcodec/ratecontrol.c index 59f69a27a8..ea6aafd098 100644 --- a/libavcodec/ratecontrol.c +++ b/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); diff --git a/libavcodec/resample.c b/libavcodec/resample.c index dfaad66216..f9502880e0 100644 --- a/libavcodec/resample.c +++ b/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; } } diff --git a/libavcodec/rv10.c b/libavcodec/rv10.c index db39f4627b..e10cbbf773 100644 --- a/libavcodec/rv10.c +++ b/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; diff --git a/libavcodec/shorten.c b/libavcodec/shorten.c index cbfa23a4e5..8e66928db5 100644 --- a/libavcodec/shorten.c +++ b/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--; } diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c index 415012f4e4..45096322f7 100644 --- a/libavcodec/vorbisdec.c +++ b/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; diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c index 4989cab162..3c1ca48f9f 100644 --- a/libavcodec/wmaprodec.c +++ b/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 */ diff --git a/libavformat/dvenc.c b/libavformat/dvenc.c index a132edbadf..0841d0e636 100644 --- a/libavformat/dvenc.c +++ b/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_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 */ AVTimecode tc; /* timecode context */ }; diff --git a/libavformat/rtpdec_jpeg.c b/libavformat/rtpdec_jpeg.c index 447dd361bc..391ae12cb3 100644 --- a/libavformat/rtpdec_jpeg.c +++ b/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; } diff --git a/libavformat/smoothstreamingenc.c b/libavformat/smoothstreamingenc.c index e51d088c48..096bf79c30 100644 --- a/libavformat/smoothstreamingenc.c +++ b/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; diff --git a/libavformat/spdifenc.c b/libavformat/spdifenc.c index bb0c363e89..cd9a7d4052 100644 --- a/libavformat/spdifenc.c +++ b/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; } diff --git a/libavresample/avresample-test.c b/libavresample/avresample-test.c index ab49e489cd..81e9bf0f50 100644 --- a/libavresample/avresample-test.c +++ b/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; diff --git a/libswscale/ppc/yuv2yuv_altivec.c b/libswscale/ppc/yuv2yuv_altivec.c index 60d50a7baa..792deb9ee7 100644 --- a/libswscale/ppc/yuv2yuv_altivec.c +++ b/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 diff --git a/libswscale/swscale.c b/libswscale/swscale.c index a4229604b3..ec42440d97 100644 --- a/libswscale/swscale.c +++ b/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) { diff --git a/tests/audiogen.c b/tests/audiogen.c index 07f0be32eb..09cf429a71 100644 --- a/tests/audiogen.c +++ b/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; diff --git a/tools/patcheck b/tools/patcheck index ca7107f5eb..8cdd4aa5b8 100755 --- a/tools/patcheck +++ b/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' |