mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
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>
This commit is contained in:
Michael Niedermayer
commit
d27edc038a
2
configure
vendored
2
configure
vendored
@ -1460,7 +1460,7 @@ HAVE_LIST="
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xmm_clobbers
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"
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# options emitted with CONFIG_ prefix but not available on command line
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# options emitted with CONFIG_ prefix but not available on the command line
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CONFIG_EXTRA="
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aandcttables
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ac3dsp
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@ -288,7 +288,7 @@ TYPEDEF_HIDES_STRUCT = NO
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# causing a significant performance penality.
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# If the system has enough physical memory increasing the cache will improve the
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# performance by keeping more symbols in memory. Note that the value works on
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# a logarithmic scale so increasing the size by one will rougly double the
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# a logarithmic scale so increasing the size by one will roughly double the
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# memory usage. The cache size is given by this formula:
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# 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0,
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# corresponding to a cache size of 2^16 = 65536 symbols
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@ -170,7 +170,7 @@ For exported names, each library has its own prefixes. Just check the existing
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code and name accordingly.
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@end itemize
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@subsection Miscellanous conventions
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@subsection Miscellaneous conventions
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@itemize @bullet
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@item
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fprintf and printf are forbidden in libavformat and libavcodec,
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@ -23,7 +23,7 @@ Let's consider the problem of minimizing:
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rate is the filesize
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distortion is the quality
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lambda is a fixed value choosen as a tradeoff between quality and filesize
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lambda is a fixed value chosen as a tradeoff between quality and filesize
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Is this equivalent to finding the best quality for a given max
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filesize? The answer is yes. For each filesize limit there is some lambda
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factor for which minimizing above will get you the best quality (using your
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@ -85,8 +85,8 @@ here are some edges we could choose from:
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/ \
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O-----2--4--O
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Finding the new best pathes and scores for each point of our new column is
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trivial given we know the previous column best pathes and scores:
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Finding the new best paths and scores for each point of our new column is
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trivial given we know the previous column best paths and scores:
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O-----0-----8
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\
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@ -842,7 +842,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
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cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
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// explicit check needed as memcpy below might not catch a NULL
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if (!cfrm->data) {
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av_log(f->avctx, AV_LOG_ERROR, "realloc falure\n");
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av_log(f->avctx, AV_LOG_ERROR, "realloc failure\n");
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return -1;
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}
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@ -597,7 +597,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
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for (w = 0; w < wi->num_windows*16; w += 16) {
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AacPsyBand *bands = &pch->band[w];
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//5.4.2.3 "Spreading" & 5.4.3 "Spreaded Energy Calculation"
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/* 5.4.2.3 "Spreading" & 5.4.3 "Spread Energy Calculation" */
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spread_en[0] = bands[0].energy;
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for (g = 1; g < num_bands; g++) {
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bands[g].thr = FFMAX(bands[g].thr, bands[g-1].thr * coeffs[g].spread_hi[0]);
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@ -617,7 +617,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
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band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
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PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
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/* 5.6.1.3.1 "Prepatory steps of the perceptual entropy calculation" */
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/* 5.6.1.3.1 "Preparatory steps of the perceptual entropy calculation" */
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pe += calc_pe_3gpp(band);
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a += band->pe_const;
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active_lines += band->active_lines;
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@ -546,7 +546,7 @@ static void decode_transform_coeffs(AC3DecodeContext *s, int blk)
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for (ch = 1; ch <= s->channels; ch++) {
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/* transform coefficients for full-bandwidth channel */
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decode_transform_coeffs_ch(s, blk, ch, &m);
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/* tranform coefficients for coupling channel come right after the
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/* transform coefficients for coupling channel come right after the
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coefficients for the first coupled channel*/
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if (s->channel_in_cpl[ch]) {
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if (!got_cplchan) {
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@ -659,7 +659,7 @@ static void count_frame_bits_fixed(AC3EncodeContext *s)
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* bit allocation parameters do not change between blocks
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* no delta bit allocation
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* no skipped data
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* no auxilliary data
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* no auxiliary data
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* no E-AC-3 metadata
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*/
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@ -65,7 +65,7 @@ void ff_acelp_filter_init_mips(ACELPFContext *c);
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* the coefficients are scaled by 2^15.
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* This array only contains the right half of the filter.
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* This filter is likely identical to the one used in G.729, though this
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* could not be determined from the original comments with certainity.
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* could not be determined from the original comments with certainty.
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*/
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extern const int16_t ff_acelp_interp_filter[61];
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@ -172,7 +172,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
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table[i][0] = -1; //codes
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}
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/* first pass: map codes and compute auxillary table sizes */
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/* first pass: map codes and compute auxiliary table sizes */
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for (i = 0; i < nb_codes; i++) {
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n = codes[i].bits;
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code = codes[i].code;
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@ -757,7 +757,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame, AVPac
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} else {
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if (!f->key_frame_ok) {
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av_log(avctx, AV_LOG_ERROR,
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"Cant decode non keyframe without valid keyframe\n");
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"Cannot decode non-keyframe without valid keyframe\n");
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return AVERROR_INVALIDDATA;
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}
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p->key_frame = 0;
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@ -644,7 +644,7 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
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}
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/* Now FLX is strange, in that it is "byte" as opposed to "pixel" run length compressed.
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* This does not give us any good oportunity to perform word endian conversion
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* This does not give us any good opportunity to perform word endian conversion
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* during decompression. So if it is required (i.e., this is not a LE target, we do
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* a second pass over the line here, swapping the bytes.
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*/
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@ -86,7 +86,7 @@ static void fill_colmap(H264Context *h, int map[2][16+32], int list, int field,
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if (!interl)
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poc |= 3;
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else if( interl && (poc&3) == 3) //FIXME store all MBAFF references so this isnt needed
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else if( interl && (poc&3) == 3) // FIXME: store all MBAFF references so this is not needed
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poc= (poc&~3) + rfield + 1;
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for(j=start; j<end; j++){
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@ -235,7 +235,7 @@
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/**
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* Pack two delta values (a,b) into one 16bit word
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* according with endianess of the host machine.
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* according with endianness of the host machine.
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*/
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#if HAVE_BIGENDIAN
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#define PD(a,b) (((a) << 8) + (b))
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@ -282,7 +282,7 @@ static const int16_t delta_tab_3_5[79] = { TAB_3_5 };
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/**
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* Pack four delta values (a,a,b,b) into one 32bit word
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* according with endianess of the host machine.
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* according with endianness of the host machine.
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*/
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#if HAVE_BIGENDIAN
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#define PD(a,b) (((a) << 24) + ((a) << 16) + ((b) << 8) + (b))
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@ -198,7 +198,7 @@ static int lag_read_prob_header(lag_rac *rac, GetBitContext *gb)
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}
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/* Comment from reference source:
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* if (b & 0x80 == 0) { // order of operations is 'wrong'; it has been left this way
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* // since the compression change is negligable and fixing it
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* // since the compression change is negligible and fixing it
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* // breaks backwards compatibility
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* b =- (signed int)b;
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* b &= 0xFF;
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@ -257,7 +257,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_BANDWIDTH,
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avctx->cutoff)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwith to %d: %s\n",
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av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwidth to %d: %s\n",
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avctx->cutoff, aac_get_error(err));
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goto error;
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}
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@ -341,7 +341,7 @@ static int encode_frame(AVCodecContext* avc_context, AVPacket *pkt,
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memcpy(pkt->data, o_packet.packet, o_packet.bytes);
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// HACK: assumes no encoder delay, this is true until libtheora becomes
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// multithreaded (which will be disabled unless explictly requested)
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// multithreaded (which will be disabled unless explicitly requested)
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pkt->pts = pkt->dts = frame->pts;
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avc_context->coded_frame->key_frame = !(o_packet.granulepos & h->keyframe_mask);
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if (avc_context->coded_frame->key_frame)
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@ -95,7 +95,7 @@ void ff_fetch_timestamp(AVCodecParserContext *s, int off, int remove){
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if ( s->cur_offset + off >= s->cur_frame_offset[i]
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&& (s->frame_offset < s->cur_frame_offset[i] ||
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(!s->frame_offset && !s->next_frame_offset)) // first field/frame
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//check is disabled because mpeg-ts doesn't send complete PES packets
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// check disabled since MPEG-TS does not send complete PES packets
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&& /*s->next_frame_offset + off <*/ s->cur_frame_end[i]){
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s->dts= s->cur_frame_dts[i];
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s->pts= s->cur_frame_pts[i];
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@ -372,7 +372,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *pkt,
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int pass;
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for(pass = 0; pass < NB_PASSES; pass++) {
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/* NOTE: a pass is completely omited if no pixels would be
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/* NOTE: a pass is completely omitted if no pixels would be
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output */
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pass_row_size = ff_png_pass_row_size(pass, bits_per_pixel, avctx->width);
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if (pass_row_size > 0) {
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@ -816,7 +816,7 @@ static int init_pass2(MpegEncContext *s)
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AVCodecContext *a= s->avctx;
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int i, toobig;
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double fps= get_fps(s->avctx);
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double complexity[5]={0,0,0,0,0}; // aproximate bits at quant=1
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double complexity[5]={0,0,0,0,0}; // approximate bits at quant=1
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uint64_t const_bits[5]={0,0,0,0,0}; // quantizer independent bits
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uint64_t all_const_bits;
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uint64_t all_available_bits= (uint64_t)(s->bit_rate*(double)rcc->num_entries/fps);
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@ -406,7 +406,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
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if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
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ibuf, istride, nb_samples1 * s->output_channels) < 0) {
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av_log(s->resample_context, AV_LOG_ERROR,
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"Audio sample format convertion failed\n");
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"Audio sample format conversion failed\n");
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return 0;
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}
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}
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@ -740,7 +740,7 @@ static int rv10_decode_frame(AVCodecContext *avctx,
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*got_frame = 1;
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ff_print_debug_info(s, pict);
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}
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s->current_picture_ptr= NULL; //so we can detect if frame_end wasnt called (find some nicer solution...)
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s->current_picture_ptr= NULL; // so we can detect if frame_end was not called (find some nicer solution...)
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}
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return avpkt->size;
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@ -526,7 +526,8 @@ static int shorten_decode_frame(AVCodecContext *avctx, void *data,
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/* get Rice code for residual decoding */
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if (cmd != FN_ZERO) {
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residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE);
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/* this is a hack as version 0 differed in definition of get_sr_golomb_shorten */
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/* This is a hack as version 0 differed in the definition
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* of get_sr_golomb_shorten(). */
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if (s->version == 0)
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residual_size--;
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}
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@ -1235,7 +1235,7 @@ static int vorbis_floor1_decode(vorbis_context *vc,
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if (highroom < lowroom) {
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room = highroom * 2;
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} else {
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room = lowroom * 2; // SPEC mispelling
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room = lowroom * 2; // SPEC misspelling
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}
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if (val) {
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floor1_flag[low_neigh_offs] = 1;
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@ -1099,7 +1099,7 @@ static int decode_subframe(WMAProDecodeCtx *s)
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s->channels_for_cur_subframe = 0;
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for (i = 0; i < s->avctx->channels; i++) {
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const int cur_subframe = s->channel[i].cur_subframe;
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/** substract already processed samples */
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/** subtract already processed samples */
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total_samples -= s->channel[i].decoded_samples;
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/** and count if there are multiple subframes that match our profile */
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@ -51,9 +51,9 @@ struct DVMuxContext {
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AVFifoBuffer *audio_data[2]; /* FIFO for storing excessive amounts of PCM */
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int frames; /* current frame number */
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int64_t start_time; /* recording start time */
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int has_audio; /* frame under contruction has audio */
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int has_video; /* frame under contruction has video */
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uint8_t frame_buf[DV_MAX_FRAME_SIZE]; /* frame under contruction */
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int has_audio; /* frame under construction has audio */
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int has_video; /* frame under construction has video */
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uint8_t frame_buf[DV_MAX_FRAME_SIZE]; /* frame under construction */
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AVTimecode tc; /* timecode context */
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};
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@ -370,7 +370,7 @@ static int jpeg_parse_packet(AVFormatContext *ctx, PayloadContext *jpeg,
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/* Prepare the JPEG packet. */
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if ((ret = ff_rtp_finalize_packet(pkt, &jpeg->frame, st->index)) < 0) {
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av_log(ctx, AV_LOG_ERROR,
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"Error occured when getting frame buffer.\n");
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"Error occurred when getting frame buffer.\n");
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return ret;
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}
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@ -51,7 +51,7 @@ typedef struct {
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char dirname[1024];
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uint8_t iobuf[32768];
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URLContext *out; // Current output stream where all output is written
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URLContext *out2; // Auxillary output stream where all output also is written
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URLContext *out2; // Auxiliary output stream where all output is also written
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URLContext *tail_out; // The actual main output stream, if we're currently seeked back to write elsewhere
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int64_t tail_pos, cur_pos, cur_start_pos;
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int packets_written;
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@ -339,7 +339,7 @@ static int spdif_header_mpeg(AVFormatContext *s, AVPacket *pkt)
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ctx->data_type = mpeg_data_type [version & 1][layer];
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ctx->pkt_offset = spdif_mpeg_pkt_offset[version & 1][layer];
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}
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// TODO Data type dependant info (normal/karaoke, dynamic range control)
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// TODO Data type dependent info (normal/karaoke, dynamic range control)
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return 0;
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}
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@ -100,7 +100,7 @@ static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
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a += M_PI * 1000.0 * 2.0 / sample_rate;
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}
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/* 1 second of varing frequency between 100 and 10000 Hz */
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/* 1 second of varying frequency between 100 and 10000 Hz */
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a = 0;
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for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
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v = sin(a) * 0.30;
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@ -1,5 +1,5 @@
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||||
/*
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* AltiVec-enhanced yuv-to-yuv convertion routines.
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||||
* AltiVec-enhanced yuv-to-yuv conversion routines.
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||||
*
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||||
* Copyright (C) 2004 Romain Dolbeau <romain@dolbeau.org>
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* based on the equivalent C code in swscale.c
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||||
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||||
@ -148,7 +148,7 @@ static void hScale8To19_c(SwsContext *c, int16_t *_dst, int dstW,
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||||
}
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||||
}
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// FIXME all pal and rgb srcFormats could do this convertion as well
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// FIXME all pal and rgb srcFormats could do this conversion as well
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// FIXME all scalers more complex than bilinear could do half of this transform
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static void chrRangeToJpeg_c(int16_t *dstU, int16_t *dstV, int width)
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||||
{
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@ -189,7 +189,7 @@ int main(int argc, char **argv)
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a += (1000 * FRAC_ONE) / sample_rate;
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}
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/* 1 second of varing frequency between 100 and 10000 Hz */
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/* 1 second of varying frequency between 100 and 10000 Hz */
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a = 0;
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for (i = 0; i < 1 * sample_rate; i++) {
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v = (int_cos(a) * 10000) >> FRAC_BITS;
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||||
@ -158,7 +158,7 @@ cat $* | tr '\n' '@' | $EGREP --color=always -o '[^a-zA-Z0-9_]([a-zA-Z0-9_]*) *=
|
||||
cat $TMP | tr '@' '\n'
|
||||
|
||||
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||||
# doesnt work
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||||
# 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'
|
||||
|
||||
|
||||
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