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Merge commit '511cf612ac979f536fd65e14603a87ca5ad435f3'

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* 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
2012-12-21 17:32:52 +01:00
parent a41bf09d9c 511cf612ac
commit d27edc038a
35 changed files with 41 additions and 40 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
configure vendored
View File

@@ -1460,7 +1460,7 @@ HAVE_LIST="
xmm_clobbers
"
# options emitted with CONFIG_ prefix but not available on command line
# options emitted with CONFIG_ prefix but not available on the command line
CONFIG_EXTRA="
aandcttables
ac3dsp

2
doc/Doxyfile
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@@ -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

2
doc/developer.texi
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@@ -170,7 +170,7 @@ For exported names, each library has its own prefixes. Just check the existing
code and name accordingly.
@end itemize
@subsection Miscellanous conventions
@subsection Miscellaneous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,

2
doc/rate_distortion.txt
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@@ -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

4
doc/viterbi.txt
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@@ -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
\

2
libavcodec/4xm.c
View File

@@ -842,7 +842,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
// explicit check needed as memcpy below might not catch a NULL
if (!cfrm->data) {
av_log(f->avctx, AV_LOG_ERROR, "realloc falure\n");
av_log(f->avctx, AV_LOG_ERROR, "realloc failure\n");
return -1;
}

4
libavcodec/aacpsy.c
View File

@@ -597,7 +597,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
for (w = 0; w < wi->num_windows*16; w += 16) {
AacPsyBand *bands = &pch->band[w];
//5.4.2.3 "Spreading" & 5.4.3 "Spreaded Energy Calculation"
/* 5.4.2.3 "Spreading" & 5.4.3 "Spread Energy Calculation" */
spread_en[0] = bands[0].energy;
for (g = 1; g < num_bands; g++) {
bands[g].thr = FFMAX(bands[g].thr, bands[g-1].thr * coeffs[g].spread_hi[0]);
@@ -617,7 +617,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
/* 5.6.1.3.1 "Prepatory steps of the perceptual entropy calculation" */
/* 5.6.1.3.1 "Preparatory steps of the perceptual entropy calculation" */
pe += calc_pe_3gpp(band);
a += band->pe_const;
active_lines += band->active_lines;

2
libavcodec/ac3dec.c
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@@ -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) {

2
libavcodec/ac3enc.c
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@@ -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
*/

2
libavcodec/acelp_filters.h
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@@ -65,7 +65,7 @@ void ff_acelp_filter_init_mips(ACELPFContext *c);
* the coefficients are scaled by 2^15.
* This array only contains the right half of the filter.
* This filter is likely identical to the one used in G.729, though this
* could not be determined from the original comments with certainity.
* could not be determined from the original comments with certainty.
*/
extern const int16_t ff_acelp_interp_filter[61];

2
libavcodec/bitstream.c
View File

@@ -172,7 +172,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
table[i][0] = -1; //codes
}
/* first pass: map codes and compute auxillary table sizes */
/* first pass: map codes and compute auxiliary table sizes */
for (i = 0; i < nb_codes; i++) {
n = codes[i].bits;
code = codes[i].code;

2
libavcodec/ffv1dec.c
View File

@@ -757,7 +757,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame, AVPac
} else {
if (!f->key_frame_ok) {
av_log(avctx, AV_LOG_ERROR,
"Cant decode non keyframe without valid keyframe\n");
"Cannot decode non-keyframe without valid keyframe\n");
return AVERROR_INVALIDDATA;
}
p->key_frame = 0;

2
libavcodec/flicvideo.c
View File

@@ -644,7 +644,7 @@ static int flic_decode_frame_15_16BPP(AVCodecContext *avctx,
}
/* Now FLX is strange, in that it is "byte" as opposed to "pixel" run length compressed.
* This does not give us any good oportunity to perform word endian conversion
* This does not give us any good opportunity to perform word endian conversion
* during decompression. So if it is required (i.e., this is not a LE target, we do
* a second pass over the line here, swapping the bytes.
*/

2
libavcodec/h264_direct.c
View File

@@ -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++){

4
libavcodec/indeo3data.h
View File

@@ -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))

2
libavcodec/lagarith.c
View File

@@ -198,7 +198,7 @@ static int lag_read_prob_header(lag_rac *rac, GetBitContext *gb)
}
/* Comment from reference source:
* if (b & 0x80 == 0) { // order of operations is 'wrong'; it has been left this way
* // since the compression change is negligable and fixing it
* // since the compression change is negligible and fixing it
* // breaks backwards compatibility
* b =- (signed int)b;
* b &= 0xFF;

2
libavcodec/libfdk-aacenc.c
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@@ -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;
}

2
libavcodec/libtheoraenc.c
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@@ -341,7 +341,7 @@ static int encode_frame(AVCodecContext* avc_context, AVPacket *pkt,
memcpy(pkt->data, o_packet.packet, o_packet.bytes);
// HACK: assumes no encoder delay, this is true until libtheora becomes
// multithreaded (which will be disabled unless explictly requested)
// multithreaded (which will be disabled unless explicitly requested)
pkt->pts = pkt->dts = frame->pts;
avc_context->coded_frame->key_frame = !(o_packet.granulepos & h->keyframe_mask);
if (avc_context->coded_frame->key_frame)

2
libavcodec/parser.c
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@@ -95,7 +95,7 @@ void ff_fetch_timestamp(AVCodecParserContext *s, int off, int remove){
if ( s->cur_offset + off >= s->cur_frame_offset[i]
&& (s->frame_offset < s->cur_frame_offset[i] ||
(!s->frame_offset && !s->next_frame_offset)) // first field/frame
//check is disabled because mpeg-ts doesn't send complete PES packets
// check disabled since MPEG-TS does not send complete PES packets
&& /*s->next_frame_offset + off <*/ s->cur_frame_end[i]){
s->dts= s->cur_frame_dts[i];
s->pts= s->cur_frame_pts[i];

2
libavcodec/pngenc.c
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@@ -372,7 +372,7 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *pkt,
int pass;
for(pass = 0; pass < NB_PASSES; pass++) {
/* NOTE: a pass is completely omited if no pixels would be
/* NOTE: a pass is completely omitted if no pixels would be
output */
pass_row_size = ff_png_pass_row_size(pass, bits_per_pixel, avctx->width);
if (pass_row_size > 0) {

2
libavcodec/ratecontrol.c
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@@ -816,7 +816,7 @@ static int init_pass2(MpegEncContext *s)
AVCodecContext *a= s->avctx;
int i, toobig;
double fps= get_fps(s->avctx);
double complexity[5]={0,0,0,0,0}; // aproximate bits at quant=1
double complexity[5]={0,0,0,0,0}; // approximate bits at quant=1
uint64_t const_bits[5]={0,0,0,0,0}; // quantizer independent bits
uint64_t all_const_bits;
uint64_t all_available_bits= (uint64_t)(s->bit_rate*(double)rcc->num_entries/fps);

2
libavcodec/resample.c
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@@ -406,7 +406,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
ibuf, istride, nb_samples1 * s->output_channels) < 0) {
av_log(s->resample_context, AV_LOG_ERROR,
"Audio sample format convertion failed\n");
"Audio sample format conversion failed\n");
return 0;
}
}

2
libavcodec/rv10.c
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@@ -740,7 +740,7 @@ static int rv10_decode_frame(AVCodecContext *avctx,
*got_frame = 1;
ff_print_debug_info(s, pict);
}
s->current_picture_ptr= NULL; //so we can detect if frame_end wasnt called (find some nicer solution...)
s->current_picture_ptr= NULL; // so we can detect if frame_end was not called (find some nicer solution...)
}
return avpkt->size;

3
libavcodec/shorten.c
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@@ -526,7 +526,8 @@ static int shorten_decode_frame(AVCodecContext *avctx, void *data,
/* get Rice code for residual decoding */
if (cmd != FN_ZERO) {
residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE);
/* this is a hack as version 0 differed in definition of get_sr_golomb_shorten */
/* This is a hack as version 0 differed in the definition
* of get_sr_golomb_shorten(). */
if (s->version == 0)
residual_size--;
}

2
libavcodec/vorbisdec.c
View File

@@ -1235,7 +1235,7 @@ static int vorbis_floor1_decode(vorbis_context *vc,
if (highroom < lowroom) {
room = highroom * 2;
} else {
room = lowroom * 2; // SPEC mispelling
room = lowroom * 2; // SPEC misspelling
}
if (val) {
floor1_flag[low_neigh_offs] = 1;

2
libavcodec/wmaprodec.c
View File

@@ -1099,7 +1099,7 @@ static int decode_subframe(WMAProDecodeCtx *s)
s->channels_for_cur_subframe = 0;
for (i = 0; i < s->avctx->channels; i++) {
const int cur_subframe = s->channel[i].cur_subframe;
/** substract already processed samples */
/** subtract already processed samples */
total_samples -= s->channel[i].decoded_samples;
/** and count if there are multiple subframes that match our profile */

6
libavformat/dvenc.c
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@@ -51,9 +51,9 @@ struct DVMuxContext {
AVFifoBuffer *audio_data[2]; /* FIFO for storing excessive amounts of PCM */
int frames; /* current frame number */
int64_t start_time; /* recording start time */
int has_audio; /* frame under contruction has audio */
int has_video; /* frame under contruction has video */
uint8_t frame_buf[DV_MAX_FRAME_SIZE]; /* frame under contruction */
int has_audio; /* frame under construction has audio */
int has_video; /* frame under construction has video */
uint8_t frame_buf[DV_MAX_FRAME_SIZE]; /* frame under construction */
AVTimecode tc; /* timecode context */
};

2
libavformat/rtpdec_jpeg.c
View File

@@ -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;
}

2
libavformat/smoothstreamingenc.c
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@@ -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;

2
libavformat/spdifenc.c
View File

@@ -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;
}

2
libavresample/avresample-test.c
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@@ -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;

2
libswscale/ppc/yuv2yuv_altivec.c
View File

@@ -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

2
libswscale/swscale.c
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@@ -148,7 +148,7 @@ static void hScale8To19_c(SwsContext *c, int16_t *_dst, int dstW,
}
}
// FIXME all pal and rgb srcFormats could do this convertion as well
// FIXME all pal and rgb srcFormats could do this conversion as well
// FIXME all scalers more complex than bilinear could do half of this transform
static void chrRangeToJpeg_c(int16_t *dstU, int16_t *dstV, int width)
{

2
tests/audiogen.c
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@@ -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;

2
tools/patcheck
View File

@@ -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'