miscellaneous typo fixes
This commit is contained in:
Diego Biurrun
parent
6906b19346
commit
511cf612ac
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@ -1305,7 +1305,7 @@ 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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@ -201,7 +201,7 @@ 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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@ -300,7 +300,7 @@ The filename passed as input has the syntax:
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@var{hostname}:@var{display_number}.@var{screen_number} specifies the
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X11 display name of the screen to grab from. @var{hostname} can be
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ommitted, and defaults to "localhost". The environment variable
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omitted, and defaults to "localhost". The environment variable
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@env{DISPLAY} contains the default display name.
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@var{x_offset} and @var{y_offset} specify the offsets of the grabbed
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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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@ -796,7 +796,7 @@ 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");
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av_log(f->avctx, AV_LOG_ERROR, "realloc failure");
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return -1;
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}
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@ -592,7 +592,7 @@ 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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@ -612,7 +612,7 @@ 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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@ -32,7 +32,7 @@
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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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@ -2292,7 +2292,7 @@ typedef struct AVCodecContext {
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/**
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* ratecontrol qmin qmax limiting method
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* 0-> clipping, 1-> use a nice continous function to limit qscale wthin qmin/qmax.
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* 0-> clipping, 1-> use a nice continuous function to limit qscale wthin qmin/qmax.
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* - encoding: Set by user.
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* - decoding: unused
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*/
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@ -169,7 +169,7 @@ 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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@ -491,7 +491,7 @@ int ff_eac3_parse_header(AC3DecodeContext *s)
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s->skip_syntax = get_bits1(gbc);
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parse_spx_atten_data = get_bits1(gbc);
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/* coupling strategy occurance and coupling use per block */
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/* coupling strategy occurrence and coupling use per block */
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num_cpl_blocks = 0;
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if (s->channel_mode > 1) {
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for (blk = 0; blk < s->num_blocks; blk++) {
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@ -824,7 +824,7 @@ static int ffv1_decode_frame(AVCodecContext *avctx, void *data,
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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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@ -581,7 +581,7 @@ 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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@ -34,7 +34,7 @@
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/**
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* G.726 11bit float.
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* G.726 Standard uses rather odd 11bit floating point arithmentic for
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* numerous occasions. It's a mistery to me why they did it this way
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* numerous occasions. It's a mystery to me why they did it this way
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* instead of simply using 32bit integer arithmetic.
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*/
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typedef struct Float11 {
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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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@ -198,8 +198,8 @@ 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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* // breaks backwards compatibilty
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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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* } else {
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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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@ -338,7 +338,7 @@ 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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@ -89,7 +89,7 @@ static inline int get_block_rate(MpegEncContext * s, DCTELEM block[64], int bloc
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* @param[in,out] block MB coefficients, these will be restored
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* @param[in] dir ac prediction direction for each 8x8 block
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* @param[out] st scantable for each 8x8 block
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* @param[in] zigzag_last_index index refering to the last non zero coefficient in zigzag order
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* @param[in] zigzag_last_index index referring to the last non zero coefficient in zigzag order
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*/
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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])
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{
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@ -120,7 +120,7 @@ static inline void restore_ac_coeffs(MpegEncContext * s, DCTELEM block[6][64], c
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* @param[in,out] block MB coefficients, these will be updated if 1 is returned
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* @param[in] dir ac prediction direction for each 8x8 block
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* @param[out] st scantable for each 8x8 block
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* @param[out] zigzag_last_index index refering to the last non zero coefficient in zigzag order
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* @param[out] zigzag_last_index index referring to the last non zero coefficient in zigzag order
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*/
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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])
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{
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@ -96,7 +96,7 @@ 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 doesnt 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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@ -367,7 +367,7 @@ 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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@ -799,7 +799,7 @@ 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= 1/av_q2d(s->avctx->time_base);
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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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@ -350,7 +350,7 @@ 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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@ -706,7 +706,7 @@ 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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@ -528,7 +528,8 @@ 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 defintion 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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@ -43,7 +43,7 @@ void ff_thread_flush(AVCodecContext *avctx);
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* Returns the next available frame in picture. *got_picture_ptr
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* will be 0 if none is available.
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* The return value on success is the size of the consumed packet for
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* compatiblity with avcodec_decode_video2(). This means the decoder
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* compatibility with avcodec_decode_video2(). This means the decoder
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* has to consume the full packet.
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*
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* Parameters are the same as avcodec_decode_video2().
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@ -281,7 +281,7 @@ int ff_vda_create_decoder(struct vda_context *vda_ctx,
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#endif
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/* Each VCL NAL in the bistream sent to the decoder
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* is preceeded by a 4 bytes length header.
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* is preceded by a 4 bytes length header.
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* Change the avcC atom header if needed, to signal headers of 4 bytes. */
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if (extradata_size >= 4 && (extradata[4] & 0x03) != 0x03) {
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uint8_t *rw_extradata;
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@ -1233,7 +1233,7 @@ 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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@ -73,7 +73,7 @@ typedef struct VP8DSPContext {
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* second dimension: 0 if no vertical interpolation is needed;
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* 1 4-tap vertical interpolation filter (my & 1)
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* 2 6-tap vertical interpolation filter (!(my & 1))
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* third dimension: same as second dimention, for horizontal interpolation
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* third dimension: same as second dimension, for horizontal interpolation
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* so something like put_vp8_epel_pixels_tab[width>>3][2*!!my-(my&1)][2*!!mx-(mx&1)](..., mx, my)
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*/
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vp8_mc_func put_vp8_epel_pixels_tab[3][3][3];
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@ -533,7 +533,7 @@ static int decode_tilehdr(WMAProDecodeCtx *s)
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int c;
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/* Should never consume more than 3073 bits (256 iterations for the
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* while loop when always the minimum amount of 128 samples is substracted
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* 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 */
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
|
|
@ -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).
|
||||
*/
|
||||
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
||||
|
|
|
|||
|
|
@ -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] = {
|
||||
|
|
|
|||
|
|
@ -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.
|
||||
*
|
||||
|
|
|
|||
|
|
@ -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.
|
||||
*
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
|
|
|
|||
|
|
@ -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...
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
|
|
@ -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)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
|
|
|||
|
|
@ -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'
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue