From 511cf612ac979f536fd65e14603a87ca5ad435f3 Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Wed, 19 Dec 2012 18:48:21 +0100 Subject: [PATCH] miscellaneous typo fixes --- configure | 2 +- doc/Doxyfile | 2 +- doc/developer.texi | 2 +- doc/indevs.texi | 2 +- doc/rate_distortion.txt | 2 +- doc/viterbi.txt | 4 ++-- libavcodec/4xm.c | 2 +- libavcodec/aacpsy.c | 4 ++-- libavcodec/ac3dec.c | 2 +- libavcodec/ac3enc.c | 2 +- libavcodec/acelp_filters.h | 2 +- libavcodec/avcodec.h | 2 +- libavcodec/bitstream.c | 2 +- libavcodec/eac3dec.c | 2 +- libavcodec/ffv1dec.c | 2 +- libavcodec/flicvideo.c | 2 +- libavcodec/g726.c | 2 +- libavcodec/h264_direct.c | 2 +- libavcodec/indeo3data.h | 4 ++-- libavcodec/lagarith.c | 4 ++-- libavcodec/libfdk-aacenc.c | 2 +- libavcodec/libtheoraenc.c | 2 +- libavcodec/mpeg4videoenc.c | 4 ++-- libavcodec/parser.c | 2 +- libavcodec/pngenc.c | 2 +- libavcodec/ratecontrol.c | 2 +- libavcodec/resample.c | 2 +- libavcodec/rv10.c | 2 +- libavcodec/shorten.c | 3 ++- libavcodec/thread.h | 2 +- libavcodec/vda_h264.c | 2 +- libavcodec/vorbisdec.c | 2 +- libavcodec/vp8dsp.h | 2 +- libavcodec/wmaprodec.c | 4 ++-- libavdevice/dv1394.h | 2 +- libavformat/avformat.h | 2 +- libavformat/aviobuf.c | 2 +- libavformat/dvenc.c | 6 +++--- libavformat/hls.c | 2 +- libavformat/hlsproto.c | 2 +- libavformat/http.h | 2 +- libavformat/rtpdec_jpeg.c | 2 +- libavformat/smoothstreamingenc.c | 2 +- libavformat/spdifenc.c | 2 +- libavformat/wtv.c | 2 +- libavformat/xmv.c | 2 +- libavresample/avresample-test.c | 2 +- libswscale/ppc/yuv2yuv_altivec.c | 2 +- libswscale/swscale.c | 2 +- tests/audiogen.c | 2 +- tools/patcheck | 4 ++-- 51 files changed, 61 insertions(+), 60 deletions(-) diff --git a/configure b/configure index f099118b665f8..c357b535aac69 100755 --- a/configure +++ b/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 diff --git a/doc/Doxyfile b/doc/Doxyfile index 1a37021c5b659..3b2236cb434be 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 aff28b845e24c..682a239abb2d3 100644 --- a/doc/developer.texi +++ b/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, diff --git a/doc/indevs.texi b/doc/indevs.texi index b0ba6ac9f36a1..868329799f0ef 100644 --- a/doc/indevs.texi +++ b/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 diff --git a/doc/rate_distortion.txt b/doc/rate_distortion.txt index a7d2c878b2734..e9711c2d5ca36 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 5362a0b76564e..97825462ccf0f 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 f78a0a21b2631..66149cc3e7355 100644 --- a/libavcodec/4xm.c +++ b/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; } diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c index 42db471428fda..e4b4405144a1a 100644 --- a/libavcodec/aacpsy.c +++ b/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; diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index acefe41644851..f15bfa2a07b8f 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 6d038ef9142e2..c0acc648500a6 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 b8715d266fbf9..6a9ebd943e5aa 100644 --- a/libavcodec/acelp_filters.h +++ b/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]; diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index 29e3701e45573..d12c72b74c941 100644 --- a/libavcodec/avcodec.h +++ b/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 */ diff --git a/libavcodec/bitstream.c b/libavcodec/bitstream.c index eec2f6dcb271f..2c8692a79de71 100644 --- a/libavcodec/bitstream.c +++ b/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; diff --git a/libavcodec/eac3dec.c b/libavcodec/eac3dec.c index 639e061f5a494..3a80cb146966f 100644 --- a/libavcodec/eac3dec.c +++ b/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++) { diff --git a/libavcodec/ffv1dec.c b/libavcodec/ffv1dec.c index b1dec7de3f93b..72f255cad166e 100644 --- a/libavcodec/ffv1dec.c +++ b/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; diff --git a/libavcodec/flicvideo.c b/libavcodec/flicvideo.c index 02bfc75da426d..d2cc6cdb4131d 100644 --- a/libavcodec/flicvideo.c +++ b/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. */ diff --git a/libavcodec/g726.c b/libavcodec/g726.c index 3e313b9752d23..dbe9e022405da 100644 --- a/libavcodec/g726.c +++ b/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 { diff --git a/libavcodec/h264_direct.c b/libavcodec/h264_direct.c index bf444958bfa16..2306b975b5b72 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; jhandle, 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 e57310ac33d7c..f20fabb8d6d9c 100644 --- a/libavcodec/libtheoraenc.c +++ b/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) diff --git a/libavcodec/mpeg4videoenc.c b/libavcodec/mpeg4videoenc.c index b145eb229bfad..986cba62fa7f7 100644 --- a/libavcodec/mpeg4videoenc.c +++ b/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]) { diff --git a/libavcodec/parser.c b/libavcodec/parser.c index 0767a3495978f..6e755f6b75c0d 100644 --- a/libavcodec/parser.c +++ b/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]; diff --git a/libavcodec/pngenc.c b/libavcodec/pngenc.c index 00a800c795b3f..b20a6d6d46786 100644 --- a/libavcodec/pngenc.c +++ b/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) { diff --git a/libavcodec/ratecontrol.c b/libavcodec/ratecontrol.c index 2cb5eeaefe589..e0b6e9bf0bed8 100644 --- a/libavcodec/ratecontrol.c +++ b/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); diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 20d7078113f16..1b3bb834f3b73 100644 --- a/libavcodec/resample.c +++ b/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; } } diff --git a/libavcodec/rv10.c b/libavcodec/rv10.c index 73af3622e6be0..9239cf7d94f0e 100644 --- a/libavcodec/rv10.c +++ b/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; diff --git a/libavcodec/shorten.c b/libavcodec/shorten.c index fad69b8d08ad9..1dc010f441d57 100644 --- a/libavcodec/shorten.c +++ b/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--; } diff --git a/libavcodec/thread.h b/libavcodec/thread.h index 782c03cbcf962..99b0ce146a476 100644 --- a/libavcodec/thread.h +++ b/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(). diff --git a/libavcodec/vda_h264.c b/libavcodec/vda_h264.c index 2a78aac61a8a0..34fcd3c6e1c47 100644 --- a/libavcodec/vda_h264.c +++ b/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; diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c index b30e614c2f4b8..aac9019ed6837 100644 --- a/libavcodec/vorbisdec.c +++ b/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; diff --git a/libavcodec/vp8dsp.h b/libavcodec/vp8dsp.h index 62cc0109899b9..bce0062c51d61 100644 --- a/libavcodec/vp8dsp.h +++ b/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]; diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c index 321c25d9f1928..e19c3d36b9654 100644 --- a/libavcodec/wmaprodec.c +++ b/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 */ diff --git a/libavdevice/dv1394.h b/libavdevice/dv1394.h index fc4df2403299c..9710ff56ea9d1 100644 --- a/libavdevice/dv1394.h +++ b/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 diff --git a/libavformat/avformat.h b/libavformat/avformat.h index 51635c4b84cb6..149b66f1c9d50 100644 --- a/libavformat/avformat.h +++ b/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). */ diff --git a/libavformat/aviobuf.c b/libavformat/aviobuf.c index b762d10a2a8cf..0da1e0579b499 100644 --- a/libavformat/aviobuf.c +++ b/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; diff --git a/libavformat/dvenc.c b/libavformat/dvenc.c index 27a444ea1ffef..a991cc6b0c239 100644 --- a/libavformat/dvenc.c +++ b/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] = { diff --git a/libavformat/hls.c b/libavformat/hls.c index 1f6b7d56ed35b..4c0e0c0b63358 100644 --- a/libavformat/hls.c +++ b/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. * diff --git a/libavformat/hlsproto.c b/libavformat/hlsproto.c index 179bdf1967a50..b750501c4d1be 100644 --- a/libavformat/hlsproto.c +++ b/libavformat/hlsproto.c @@ -36,7 +36,7 @@ * 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. * diff --git a/libavformat/http.h b/libavformat/http.h index 3579ad745ac6e..f0d9d4aea8f7a 100644 --- a/libavformat/http.h +++ b/libavformat/http.h @@ -40,7 +40,7 @@ void ff_http_init_auth_state(URLContext *dest, const URLContext *src); * * @param h pointer to the ressource * @param uri uri used to perform the request - * @return a negative value if an error condition occured, 0 + * @return a negative value if an error condition occurred, 0 * otherwise */ int ff_http_do_new_request(URLContext *h, const char *uri); diff --git a/libavformat/rtpdec_jpeg.c b/libavformat/rtpdec_jpeg.c index 9f73f7d5dcfd9..25bb88d0d1b7c 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 1ed675a27212c..d26af0564b422 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 77af92e1f33ab..dcdabae1de364 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/libavformat/wtv.c b/libavformat/wtv.c index 7bb421b0ce29c..2e5d39cff2e48 100644 --- a/libavformat/wtv.c +++ b/libavformat/wtv.c @@ -221,7 +221,7 @@ static AVIOContext * wtvfile_open_sector(int first_sector, uint64_t length, int } wf->length = length; - /* seek to intial sector */ + /* seek to initial sector */ wf->position = 0; if (avio_seek(s->pb, (int64_t)wf->sectors[0] << WTV_SECTOR_BITS, SEEK_SET) < 0) { av_free(wf->sectors); diff --git a/libavformat/xmv.c b/libavformat/xmv.c index 5041c571c3738..3f926eff9c04d 100644 --- a/libavformat/xmv.c +++ b/libavformat/xmv.c @@ -298,7 +298,7 @@ static int xmv_process_packet_header(AVFormatContext *s) * short for every audio track. But as playing around with XMV files with * ADPCM audio showed, taking the extra 4 bytes from the audio data gives * you either completely distorted audio or click (when skipping the - * remaining 68 bytes of the ADPCM block). Substracting 4 bytes for every + * remaining 68 bytes of the ADPCM block). Subtracting 4 bytes for every * audio track from the video data works at least for the audio. Probably * some alignment thing? * The video data has (always?) lots of padding, so it should work out... diff --git a/libavresample/avresample-test.c b/libavresample/avresample-test.c index ab49e489cdebe..81e9bf0f50684 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 e68cccf6a76fc..5aa1820351b5e 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 * based on the equivalent C code in swscale.c diff --git a/libswscale/swscale.c b/libswscale/swscale.c index c1920de0a6420..dac8b37468965 100644 --- a/libswscale/swscale.c +++ b/libswscale/swscale.c @@ -163,7 +163,7 @@ 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 acb380da507da..4fa465638a0f7 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 78ca8246f72fd..d22cf3c5aad81 100755 --- a/tools/patcheck +++ b/tools/patcheck @@ -19,7 +19,7 @@ echo This tool is intended to help a human check/review patches it is very far f echo being free of false positives and negatives, its output are just hints of what echo may or may not be bad. When you use it and it misses something or detects echo something wrong, fix it and send a patch to the libav-devel mailing list. -echo License:GPL Autor: Michael Niedermayer +echo License:GPL Author: Michael Niedermayer ERE_PRITYP='(unsigned *|)(char|short|long|int|long *int|short *int|void|float|double|(u|)int(8|16|32|64)_t)' ERE_TYPES='(const|static|av_cold|inline| *)*('$ERE_PRITYP'|[a-zA-Z][a-zA-Z0-9_]*)[* ]{1,}[a-zA-Z][a-zA-Z0-9_]*' @@ -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'