diff options
51 files changed, 61 insertions, 60 deletions
@@ -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 1a37021c5b..3b2236cb43 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 aff28b845e..682a239abb 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 b0ba6ac9f3..868329799f 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 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 f78a0a21b2..66149cc3e7 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 42db471428..e4b4405144 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 acefe41644..f15bfa2a07 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 6d038ef914..c0acc64850 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 b8715d266f..6a9ebd943e 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 29e3701e45..d12c72b74c 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 eec2f6dcb2..2c8692a79d 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 639e061f5a..3a80cb1469 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 b1dec7de3f..72f255cad1 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 02bfc75da4..d2cc6cdb41 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 3e313b9752..dbe9e02240 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 bf444958bf..2306b975b5 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 2ef8ea74bd..28c9bb6061 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 7d7a1a7ff1..33dd8b0c53 100644 --- a/libavcodec/lagarith.c +++ 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 { diff --git a/libavcodec/libfdk-aacenc.c b/libavcodec/libfdk-aacenc.c index 85ccb10148..00397d58ed 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 e57310ac33..f20fabb8d6 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 b145eb229b..986cba62fa 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 0767a34959..6e755f6b75 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 00a800c795..b20a6d6d46 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 2cb5eeaefe..e0b6e9bf0b 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 20d7078113..1b3bb834f3 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 73af3622e6..9239cf7d94 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 fad69b8d08..1dc010f441 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 782c03cbcf..99b0ce146a 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 2a78aac61a..34fcd3c6e1 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 b30e614c2f..aac9019ed6 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 62cc010989..bce0062c51 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 321c25d9f1..e19c3d36b9 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 fc4df24032..9710ff56ea 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 51635c4b84..149b66f1c9 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 b762d10a2a..0da1e0579b 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 27a444ea1f..a991cc6b0c 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 1f6b7d56ed..4c0e0c0b63 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 179bdf1967..b750501c4d 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 3579ad745a..f0d9d4aea8 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 9f73f7d5dc..25bb88d0d1 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 1ed675a272..d26af0564b 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 77af92e1f3..dcdabae1de 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 7bb421b0ce..2e5d39cff2 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 5041c571c3..3f926eff9c 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 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 e68cccf6a7..5aa1820351 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 c1920de0a6..dac8b37468 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 acb380da50..4fa465638a 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 78ca8246f7..d22cf3c5aa 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' |