[FFmpeg-devel] [PATCH] avformat/hlsenc: second_levels flags process function extract

[Steven Liu]

the SECOND_LEVEL* flags process and name is too long
extract all of them output to funtions, make code clear

Signed-off-by: Steven Liu 
---
 libavformat/hlsenc.c | 239 +--
 1 file changed, 136 insertions(+), 103 deletions(-)

diff --git a/libavformat/hlsenc.c b/libavformat/hlsenc.c
index b8122f1..5df2514 100644
--- a/libavformat/hlsenc.c
+++ b/libavformat/hlsenc.c
@@ -470,17 +470,9 @@ static HLSSegment *find_segment_by_filename(HLSSegment 
*segment, const char *fil
 return (HLSSegment *) NULL;
 }
 
-/* Create a new segment and append it to the segment list */
-static int hls_append_segment(struct AVFormatContext *s, HLSContext *hls, 
double duration,
-  int64_t pos, int64_t size)
+static int sls_flags_filename_process(struct AVFormatContext *s, HLSContext 
*hls, HLSSegment *en, double duration,
+ int64_t pos, int64_t size)
 {
-HLSSegment *en = av_malloc(sizeof(*en));
-const char  *filename;
-int ret;
-
-if (!en)
-return AVERROR(ENOMEM);
-
 if ((hls->flags & (HLS_SECOND_LEVEL_SEGMENT_SIZE | 
HLS_SECOND_LEVEL_SEGMENT_DURATION)) &&
 strlen(hls->current_segment_final_filename_fmt)) {
 av_strlcpy(hls->avf->filename, 
hls->current_segment_final_filename_fmt, sizeof(hls->avf->filename));
@@ -521,7 +513,127 @@ static int hls_append_segment(struct AVFormatContext *s, 
HLSContext *hls, double
 av_free(filename);
 }
 }
+return 0;
+}
+
+static int sls_flag_check_duration_size_index(HLSContext *hls)
+{
+int ret = 0;
+
+if (hls->flags & HLS_SECOND_LEVEL_SEGMENT_DURATION) {
+ av_log(hls, AV_LOG_ERROR,
+"second_level_segment_duration hls_flag requires use_localtime 
to be true\n");
+ ret = AVERROR(EINVAL);
+}
+if (hls->flags & HLS_SECOND_LEVEL_SEGMENT_SIZE) {
+ av_log(hls, AV_LOG_ERROR,
+"second_level_segment_size hls_flag requires use_localtime to 
be true\n");
+ ret = AVERROR(EINVAL);
+}
+if (hls->flags & HLS_SECOND_LEVEL_SEGMENT_INDEX) {
+av_log(hls, AV_LOG_ERROR,
+   "second_level_segment_index hls_flag requires use_localtime to 
be true\n");
+ret = AVERROR(EINVAL);
+}
+
+return ret;
+}
+
+static int sls_flag_check_duration_size(HLSContext *hls)
+{
+const char *proto = avio_find_protocol_name(hls->basename);
+int segment_renaming_ok = proto && !strcmp(proto, "file");
+int ret = 0;
+
+if ((hls->flags & HLS_SECOND_LEVEL_SEGMENT_DURATION) && 
!segment_renaming_ok) {
+ av_log(hls, AV_LOG_ERROR,
+"second_level_segment_duration hls_flag works only with file 
protocol segment names\n");
+ ret = AVERROR(EINVAL);
+}
+if ((hls->flags & HLS_SECOND_LEVEL_SEGMENT_SIZE) && !segment_renaming_ok) {
+ av_log(hls, AV_LOG_ERROR,
+"second_level_segment_size hls_flag works only with file 
protocol segment names\n");
+ ret = AVERROR(EINVAL);
+}
+
+return ret;
+}
+
+static void sls_flag_file_rename(HLSContext *hls, char *old_filename) {
+if ((hls->flags & (HLS_SECOND_LEVEL_SEGMENT_SIZE | 
HLS_SECOND_LEVEL_SEGMENT_DURATION)) &&
+strlen(hls->current_segment_final_filename_fmt)) {
+ff_rename(old_filename, hls->avf->filename, hls);
+}
+}
+
+static int sls_flag_use_localtime_filename(AVFormatContext *oc, HLSContext *c)
+{
+if (c->flags & HLS_SECOND_LEVEL_SEGMENT_INDEX) {
+char * filename = av_strdup(oc->filename);  // %%d will be %d after 
strftime
+if (!filename)
+return AVERROR(ENOMEM);
+if (replace_int_data_in_filename(oc->filename, sizeof(oc->filename),
+#if FF_API_HLS_WRAP
+filename, 'd', c->wrap ? c->sequence % c->wrap : c->sequence) < 1) 
{
+#else
+filename, 'd', c->sequence) < 1) {
+#endif
+av_log(c, AV_LOG_ERROR, "Invalid second level segment filename 
template '%s', "
+"you can try to remove second_level_segment_index flag\n",
+   filename);
+av_free(filename);
+return AVERROR(EINVAL);
+}
+av_free(filename);
+}
+if (c->flags & (HLS_SECOND_LEVEL_SEGMENT_SIZE | 
HLS_SECOND_LEVEL_SEGMENT_DURATION)) {
+av_strlcpy(c->current_segment_final_filename_fmt, oc->filename,
+   sizeof(c->current_segment_final_filename_fmt));
+if (c->flags & HLS_SECOND_LEVEL_SEGMENT_SIZE) {
+char * filename = av_strdup(oc->filename);  // %%s will be %s 
after strftime
+if (!filename)
+return AVERROR(ENOMEM);
+if (replace_int_data_in_filename(oc->filename, 
sizeof(oc->filename), filename, 's', 0) < 1) {
+av_log(c, AV_LOG_ERROR, "Invalid second level segment filename 
template '%s', "
+"you can try to remove second_level_segment_size 

Re: [FFmpeg-devel] [PATCH] ffprobe: add AVCodecContext help message into ffprobe

[Steven Liu]

2017-03-09 16:14 GMT+08:00 Steven Liu :

> because the ffprobe can use AVCodecContext parameters
>
> Signed-off-by: Steven Liu 
> ---
>  ffprobe.c | 1 +
>  1 file changed, 1 insertion(+)
>
> diff --git a/ffprobe.c b/ffprobe.c
> index c85c3a1..b104390 100644
> --- a/ffprobe.c
> +++ b/ffprobe.c
> @@ -3004,6 +3004,7 @@ void show_help_default(const char *opt, const char
> *arg)
>  printf("\n");
>
>  show_help_children(avformat_get_class(), AV_OPT_FLAG_DECODING_PARAM);
> +show_help_children(avcodec_get_class(), AV_OPT_FLAG_DECODING_PARAM);
>  }
>
>  /**
> --
> 2.10.1.382.ga23ca1b.dirty
>
>
>
> ___
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel



pushed


Thanks!
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[FFmpeg-devel] [PATCH 3/3] avcodec/amrwbdec: Fix runtime error: left shift of negative value -1

[Michael Niedermayer]

Fixes: 763/clusterfuzz-testcase-6007567320875008

Found-by: continuous fuzzing process 
https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer 
---
 libavcodec/amrwbdec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 999bfb99dc..57aed874cc 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -262,7 +262,7 @@ static void decode_pitch_lag_high(int *lag_int, int 
*lag_frac, int pitch_index,
 *lag_frac = pitch_index - (*lag_int << 2) + 136;
 } else if (pitch_index < 440) {
 *lag_int  = (pitch_index + 257 - 376) >> 1;
-*lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
+*lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
 /* the actual resolution is 1/2 but expressed as 1/4 */
 } else {
 *lag_int  = pitch_index - 280;
-- 
2.11.0

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[FFmpeg-devel] [PATCH 2/3] avcodec/dca_xll: Fix runtime error: signed integer overflow: 1762028192 + 698372290 cannot be represented in type 'int'

[Michael Niedermayer]

Fixes: 762/clusterfuzz-testcase-5927683747741696

Found-by: continuous fuzzing process 
https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer 
---
 libavcodec/dca_xll.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c
index 6cebda35e4..b7331e04b9 100644
--- a/libavcodec/dca_xll.c
+++ b/libavcodec/dca_xll.c
@@ -658,7 +658,7 @@ static void chs_filter_band_data(DCAXllDecoder *s, 
DCAXllChSet *c, int band)
 // Inverse fixed coefficient prediction
 for (j = 0; j < b->fixed_pred_order[i]; j++)
 for (k = 1; k < nsamples; k++)
-buf[k] += buf[k - 1];
+buf[k] += (unsigned)buf[k - 1];
 }
 }
 
-- 
2.11.0

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[FFmpeg-devel] [PATCH 1/3] avcodec/wavpack: Fix runtime error: signed integer overflow: -2147483648 + -83886075 cannot be represented in type 'int'

[Michael Niedermayer]

Fixes: 761/clusterfuzz-testcase-544252097536

Found-by: continuous fuzzing process 
https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer 
---
 libavcodec/wavpack.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/libavcodec/wavpack.h b/libavcodec/wavpack.h
index 0196574aab..445d593c3b 100644
--- a/libavcodec/wavpack.h
+++ b/libavcodec/wavpack.h
@@ -99,8 +99,8 @@ typedef struct WvChannel {
 
 // macros for manipulating median values
 #define GET_MED(n) ((c->median[n] >> 4) + 1)
-#define DEC_MED(n) c->median[n] -= ((c->median[n] + (128 >> (n)) - 2) / (128 
>> (n))) * 2
-#define INC_MED(n) c->median[n] += ((c->median[n] + (128 >> (n))) / (128 
>> (n))) * 5
+#define DEC_MED(n) c->median[n] -= ((c->median[n] + (128 >> (n)) - 2) / (128 
>> (n))) * 2U
+#define INC_MED(n) c->median[n] += ((c->median[n] + (128 >> (n))) / (128 
>> (n))) * 5U
 
 // macros for applying weight
 #define UPDATE_WEIGHT_CLIP(weight, delta, samples, in) \
-- 
2.11.0

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[FFmpeg-devel] [PATCH] avfilter/af_firequalizer: add av_restrict on convolution func

[Muhammad Faiz]

slightly improved speed

Signed-off-by: Muhammad Faiz 
---
 libavfilter/af_firequalizer.c | 8 
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/libavfilter/af_firequalizer.c b/libavfilter/af_firequalizer.c
index 5c6fd54..4243d66 100644
--- a/libavfilter/af_firequalizer.c
+++ b/libavfilter/af_firequalizer.c
@@ -197,8 +197,8 @@ static int query_formats(AVFilterContext *ctx)
 return ff_set_common_samplerates(ctx, formats);
 }
 
-static void fast_convolute(FIREqualizerContext *s, const float *kernel_buf, 
float *conv_buf,
-   OverlapIndex *idx, float *data, int nsamples)
+static void fast_convolute(FIREqualizerContext *av_restrict s, const float 
*av_restrict kernel_buf, float *av_restrict conv_buf,
+   OverlapIndex *av_restrict idx, float *av_restrict 
data, int nsamples)
 {
 if (nsamples <= s->nsamples_max) {
 float *buf = conv_buf + idx->buf_idx * s->rdft_len;
@@ -235,8 +235,8 @@ static void fast_convolute(FIREqualizerContext *s, const 
float *kernel_buf, floa
 }
 }
 
-static void fast_convolute2(FIREqualizerContext *s, const float *kernel_buf, 
FFTComplex *conv_buf,
-OverlapIndex *idx, float *data0, float *data1, int 
nsamples)
+static void fast_convolute2(FIREqualizerContext *av_restrict s, const float 
*av_restrict kernel_buf, FFTComplex *av_restrict conv_buf,
+OverlapIndex *av_restrict idx, float *av_restrict 
data0, float *av_restrict data1, int nsamples)
 {
 if (nsamples <= s->nsamples_max) {
 FFTComplex *buf = conv_buf + idx->buf_idx * s->rdft_len;
-- 
2.9.3

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Re: [FFmpeg-devel] [PATCH] avcodec/vp56: Reset have_undamaged_frame on resolution changes

[Michael Niedermayer]

On Thu, Mar 09, 2017 at 05:55:32PM +0100, Michael Niedermayer wrote:
> Fixes: timeout in 758/clusterfuzz-testcase-4720832028868608
> 
> Found-by: continuous fuzzing process 
> https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
> Signed-off-by: Michael Niedermayer 
> ---
>  libavcodec/vp56.c | 2 ++
>  1 file changed, 2 insertions(+)

applied

[...]
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Avoid a single point of failure, be that a person or equipment.


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Re: [FFmpeg-devel] [PATCH] avcodec/h264_ps: Forward errors from decode_scaling_list()

[Michael Niedermayer]

On Thu, Mar 09, 2017 at 01:54:24PM +0100, Michael Niedermayer wrote:
> Signed-off-by: Michael Niedermayer 
> ---
>  libavcodec/h264_ps.c | 44 ++--
>  1 file changed, 26 insertions(+), 18 deletions(-)

applied

[...]
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to question oneself and others. -- Socrates


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Re: [FFmpeg-devel] [PATCH 2/2] avcodec/h264_direct: Fix runtime error: signed integer overflow: 2147483647 - -14133 cannot be represented in type 'int'

[Michael Niedermayer]

On Fri, Mar 10, 2017 at 04:01:36PM +0100, wm4 wrote:
> On Fri, 10 Mar 2017 15:24:52 +0100
> Michael Niedermayer  wrote:
> 
> > Fixes: 755/clusterfuzz-testcase-5369072516595712
> > 
> > See: [FFmpeg-devel] [PATCH 1/2] avcodec/h264_direct: Fix runtime error: 
> > signed integer overflow: 2147483647 - -14133 cannot be represented in type 
> > 'int'
> > 
> > Found-by: continuous fuzzing process 
> > https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
> > Signed-off-by: Michael Niedermayer 
> > ---
> >  libavcodec/h264_direct.c | 7 ++-
> >  1 file changed, 6 insertions(+), 1 deletion(-)
> > 
> > diff --git a/libavcodec/h264_direct.c b/libavcodec/h264_direct.c
> > index cbb84665b3..66e54479d1 100644
> > --- a/libavcodec/h264_direct.c
> > +++ b/libavcodec/h264_direct.c
> > @@ -39,7 +39,12 @@ static int get_scale_factor(H264SliceContext *sl,
> >  int poc, int poc1, int i)
> >  {
> >  int poc0 = sl->ref_list[0][i].poc;
> > -int td = av_clip_int8(poc1 - poc0);
> > +int64_t pocdiff = poc1 - (int64_t)poc0;
> > +int td = av_clip_int8(pocdiff);
> > +
> > +if (pocdiff != (int)pocdiff)
> > +avpriv_request_sample(sl->h264->avctx, "pocdiff overflow\n");
> > +
> >  if (td == 0 || sl->ref_list[0][i].parent->long_ref) {
> >  return 256;
> >  } else {
> 
> Hard to image that these poc values aren't bounded by something else,
> but I don't know.

> 
> Also the previous patch didn't have this request_sample call, which
> inflates this whole thing by 5 lines of code.

yes thats why i suggested it initially.
SUINT allows overflow detection simply by  #define CHECKED 1
and running under ubsan

otherwise an excplicit check is needed to detect such occurances

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Breaking DRM is a little like attempting to break through a door even
though the window is wide open and the only thing in the house is a bunch
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Re: [FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Moritz Barsnick]

On Sat, Mar 11, 2017 at 03:36:46 +0530, Sanchit Sinha wrote:
> Subject: [PATCH] Changes to af_ambisonic, Makefile, allfilters.c and
> Changelog

This is not a proper commit message.

> ---
>  Changelog  | 1591
> 
>  libavfilter/Makefile   |  375 +++
>  libavfilter/af_ambisonic.c |  137 
>  libavfilter/allfilters.c   |  383 +++
>  4 files changed, 2486 insertions(+)
>  create mode 100644 Changelog
>  create mode 100644 libavfilter/Makefile
>  create mode 100644 libavfilter/af_ambisonic.c
>  create mode 100644 libavfilter/allfilters.c

How did you manage to make these seem like new files? Your patch is
supposed to contain the *changes* you made. Only
libavfilter/af_ambisonic.c would be added. The others are existing
files. How did you manage to create this patch?

The development process is approximately (excuse imprecisions):

git clone ...
cd ...
add your file
modify the other files
git add  
git commit
  ... write a proper commit message - see other commits for inspiration
git format-patch ...
git send-email ...

(Don't forget to check that a complete compile works with your
modifications. And that the resulting binary does the right stuff.)

The last step will especially avoid what we are seeing here in your
e-mails, that your e-mail user agent is breaking the patches by
inserting newlines.


> +AVFILTER_DEFINE_CLASS(ambisonic);
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats = NULL;
> + AVFilterChannelLayouts *layout = NULL;
> + int ret;
> + if ((ret = ff_add_format (, AV_SAMPLE_FMT_FLTP   )) < 0 ||
> + (ret = ff_set_common_formats (ctx, formats )) < 0 ||
> + (ret = ff_add_channel_layout ( , AV_CH_LAYOUT_4POINT0)) < 0 ||
> + (ret = ff_set_common_channel_layouts (ctx, layout   )) < 0)
> + return ret;
> + formats = ff_all_samplerates();
> + return ff_set_common_samplerates(ctx, formats);
> +}

Your indentation is all wrong. What happened?
http://ffmpeg.org/developer.html#Code-formatting-conventions

> +};
> \ No newline at end of file

Paul asked you to fix this (and other stuff).

Moritz
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Re: [FFmpeg-devel] [PATCH 1/2] avcodec/dca_xll: Fix runtime error: signed integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'

[Michael Niedermayer]

On Fri, Mar 10, 2017 at 06:50:34PM +0100, Hendrik Leppkes wrote:
> On Fri, Mar 10, 2017 at 3:24 PM, Michael Niedermayer
>  wrote:
> > Fixes: 732/clusterfuzz-testcase-4872990070145024
> >
> > See: [FFmpeg-devel] [PATCH 2/6] avcodec/dca_xll: Fix runtime error: signed 
> > integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'
> > Found-by: continuous fuzzing process 
> > https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
> > Signed-off-by: Michael Niedermayer 
> > ---
> >  libavcodec/dca_xll.c | 2 +-
> >  1 file changed, 1 insertion(+), 1 deletion(-)
> >
> > diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c
> > index 6cebda35e4..0fe90c7348 100644
> > --- a/libavcodec/dca_xll.c
> > +++ b/libavcodec/dca_xll.c
> > @@ -1312,7 +1312,7 @@ static int combine_residual_frame(DCAXllDecoder *s, 
> > DCAXllChSet *c)
> >  } else {
> >  // No downmix scaling
> >  for (n = 0; n < nsamples; n++)
> > -dst[n] += (src[n] + round) >> shift;
> > +dst[n] += (unsigned)((src[n] + round) >> shift);
> >  }
> >  }
> >
> 
> audio samples are typically signed, using negative values too, is this
> really safe to just cast it like that, without modifying the value in
> the end?

signed and unsigned operations for +,-,*,<< are identical on twos
complement with in and output data types matching and twos complement

(strictly speaking in the signed case some are undefined in C ...)

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Re: [FFmpeg-devel] [PATCH] Add XPM decoder

[Moritz Barsnick]

On Sat, Mar 11, 2017 at 00:34:18 +0530, Paras wrote:
>  OBJS-$(CONFIG_XBM_DECODER) += xbmdec.o
>  OBJS-$(CONFIG_XBM_ENCODER) += xbmenc.o
> +OBJS-$(CONFIG_XPM_DECODER) += xpmdec.o
>  OBJS-$(CONFIG_XFACE_DECODER)   += xfacedec.o xface.o

I do understand that this format is similar to XBM, but this code might
be needed to be ordered alphabetically, IIUC.

>  REGISTER_DECODER(XAN_WC4,   xan_wc4);
>  REGISTER_ENCDEC (XBM,   xbm);
> +REGISTER_DECODER (XPM,   xpm);
>  REGISTER_ENCDEC (XFACE, xface);

Same here (and a space too much).

> --- a/libavcodec/avcodec.h
> +++ b/libavcodec/avcodec.h
> @@ -377,6 +377,7 @@ enum AVCodecID {
>  AV_CODEC_ID_XWD,
>  AV_CODEC_ID_CDXL,
>  AV_CODEC_ID_XBM,
> +AV_CODEC_ID_XPM,
>  AV_CODEC_ID_ZEROCODEC,
>  AV_CODEC_ID_MSS1,
>  AV_CODEC_ID_MSA1,

Did you read the comment at the top of that section?

 * If you add a codec ID to this list, add it so that
 * 1. no value of an existing codec ID changes (that would break ABI),
 * 2. it is as close as possible to similar codecs


> +static int color_table_compare(const void *lhs, const void *rhs)
> +{
> +return av_strcasecmp(lhs, ((const ColorEntry *)rhs)->name);
> +}
> +
> +static const ColorEntry color_table[] = {
> +{ "AliceBlue",{ 0xF0, 0xF8, 0xFF } },
> +{ "AntiqueWhite", { 0xFA, 0xEB, 0xD7 } },
> +{ "Aqua", { 0x00, 0xFF, 0xFF } },
[...]

Is this duplicated from libavutil/parseutils.c?

> +static uint32_t hexstring_to_rgba(const char *p, int len){
> +uint32_t ret = 0xFF00;
> +const ColorEntry *entry;
> +char color_name[100];
> +
> +if(*p == '#'){
> +p++;
> +len--;
> +if (len == 3) {
> +ret |= (convert(p[2]) <<  4) |
> +   (convert(p[1]) << 12) |
> +   (convert(p[0]) << 20);

So is this a modified or redesigned av_parse_color()? Just wondering.

> +else{

If this is new code, and not copied verbatim from somewhere else, you
need to follow coding style.

> +av_log(avctx, AV_LOG_ERROR, "invalid number of colors or chars per 
> pixel\n");

Feel free to quote actual offending values...

> +av_log(avctx, AV_LOG_ERROR, "unsupported number of chars per 
> pixel\n");
> +return AVERROR_PATCHWELCOME;

...especially when asking for a patch.

Sorry for nitpicking without looking at the actual code.

Moritz
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Re: [FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Sanchit Sinha]

Hi,
I specific changes we talked about. Also updated apart from af_ambisonic.c
: allfilters.c, Makefile and Changelog.

From b74755ae0ea6741f01d2c1b2c79056984674c331 Mon Sep 17 00:00:00 2001
From: Sanchit Sinha 
Date: Sat, 11 Mar 2017 03:31:51 +0530
Subject: [PATCH] Changes to af_ambisonic, Makefile, allfilters.c and
Changelog

---
 Changelog  | 1591

 libavfilter/Makefile   |  375 +++
 libavfilter/af_ambisonic.c |  137 
 libavfilter/allfilters.c   |  383 +++
 4 files changed, 2486 insertions(+)
 create mode 100644 Changelog
 create mode 100644 libavfilter/Makefile
 create mode 100644 libavfilter/af_ambisonic.c
 create mode 100644 libavfilter/allfilters.c

diff --git a/Changelog b/Changelog
new file mode 100644
index 000..155c309
--- /dev/null
+++ b/Changelog
@@ -0,0 +1,1591 @@
+Entries are sorted chronologically from oldest to youngest within each
release,
+releases are sorted from youngest to oldest.
+
+version :
+- CrystalHD decoder moved to new decode API
+- add internal ebur128 library, remove external libebur128 dependency
+- Pro-MPEG CoP #3-R2 FEC protocol
+- premultiply video filter
+- Support for spherical videos
+- configure now fails if autodetect-libraries are requested but not found
+- PSD Decoder
+- 16.8 floating point pcm decoder
+- 24.0 floating point pcm decoder
+- Apple Pixlet decoder
+- QDMC audio decoder
+- NewTek SpeedHQ decoder
+- MIDI Sample Dump Standard demuxer
+- readeia608 filter
+- Sample Dump eXchange demuxer
+- abitscope multimedia filter
+- Scenarist Closed Captions demuxer and muxer
+- threshold filter
+- midequalizer filter
+- Optimal Huffman tables for (M)JPEG encoding
+- FM Screen Capture Codec decoder
+- native Opus encoder
+- ScreenPressor decoder
+- incomplete ClearVideo decoder
+- Ambisonic Decoder
+
+version 3.2:
+- libopenmpt demuxer
+- tee protocol
+- Changed metadata print option to accept general urls
+- Alias muxer for Ogg Video (.ogv)
+- VP8 in Ogg muxing
+- curves filter doesn't automatically insert points at x=0 and x=1 anymore
+- 16-bit support in curves filter and selectivecolor filter
+- OpenH264 decoder wrapper
+- MediaCodec H.264/HEVC/MPEG-4/VP8/VP9 hwaccel
+- True Audio (TTA) muxer
+- crystalizer audio filter
+- acrusher audio filter
+- bitplanenoise video filter
+- floating point support in als decoder
+- fifo muxer
+- maskedclamp filter
+- hysteresis filter
+- lut2 filter
+- yuvtestsrc filter
+- CUDA CUVID H.263/VP8/VP9/10 bit HEVC (Dithered) Decoding
+- vaguedenoiser filter
+- added threads option per filter instance
+- weave filter
+- gblur filter
+- avgblur filter
+- sobel and prewitt filter
+- MediaCodec HEVC/MPEG-4/VP8/VP9 decoding
+- Meridian Lossless Packing (MLP) / TrueHD encoder
+- Non-Local Means (nlmeans) denoising filter
+- sdl2 output device and ffplay support
+- sdl1 output device and sdl1 support removed
+- extended mov edit list support
+- libfaac encoder removed
+- Matroska muxer now writes CRC32 elements by default in all Level 1
elements
+- sidedata video and asidedata audio filter
+- Changed mapping of rtp MIME type G726 to codec g726le.
+- spec compliant VAAPI/DXVA2 VC-1 decoding of slices in frame-coded images
+
+
+version 3.1:
+- DXVA2-accelerated HEVC Main10 decoding
+- fieldhint filter
+- loop video filter and aloop audio filter
+- Bob Weaver deinterlacing filter
+- firequalizer filter
+- datascope filter
+- bench and abench filters
+- ciescope filter
+- protocol blacklisting API
+- MediaCodec H264 decoding
+- VC-2 HQ RTP payload format (draft v1) depacketizer and packetizer
+- VP9 RTP payload format (draft v2) packetizer
+- AudioToolbox audio decoders
+- AudioToolbox audio encoders
+- coreimage filter (GPU based image filtering on OSX)
+- libdcadec removed
+- bitstream filter for extracting DTS core
+- ADPCM IMA DAT4 decoder
+- musx demuxer
+- aix demuxer
+- remap filter
+- hash and framehash muxers
+- colorspace filter
+- hdcd filter
+- readvitc filter
+- VAAPI-accelerated format conversion and scaling
+- libnpp/CUDA-accelerated format conversion and scaling
+- Duck TrueMotion 2.0 Real Time decoder
+- Wideband Single-bit Data (WSD) demuxer
+- VAAPI-accelerated H.264/HEVC/MJPEG encoding
+- DTS Express (LBR) decoder
+- Generic OpenMAX IL encoder with support for Raspberry Pi
+- IFF ANIM demuxer & decoder
+- Direct Stream Transfer (DST) decoder
+- loudnorm filter
+- MTAF demuxer and decoder
+- MagicYUV decoder
+- OpenExr improvements (tile data and B44/B44A support)
+- BitJazz SheerVideo decoder
+- CUDA CUVID H264/HEVC decoder
+- 10-bit depth support in native utvideo decoder
+- libutvideo wrapper removed
+- YUY2 Lossless Codec decoder
+- VideoToolbox H.264 encoder
+- VAAPI-accelerated MPEG-2 and VP8 encoding
+
+
+version 3.0:
+- Common Encryption (CENC) MP4 encoding and decoding support
+- DXV decoding
+- extrastereo filter
+- ocr filter
+- alimiter filter
+- stereowiden filter
+- stereotools filter
+- rubberband 

Re: [FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Paul B Mahol]

On 3/10/17, Sanchit Sinha  wrote:
> Is the conversion okay now? I used pointers there because in one of our
> discussions you mentioned that out->ex should also be written on by
> references.(Maybe I didn't comprehend.. apologies on that). I will patch
> the makefile, allfilters.c and changelog in a while.

Its ok somehow. But pointers should be set outside of loop and acessed
in side loop like this:

w[iter] ...
x[iter] ...

and so on.
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Re: [FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Sanchit Sinha]

Is the conversion okay now? I used pointers there because in one of our
discussions you mentioned that out->ex should also be written on by
references.(Maybe I didn't comprehend.. apologies on that). I will patch
the makefile, allfilters.c and changelog in a while.
Thank You




On Sat, Mar 11, 2017 at 1:50 AM, Paul B Mahol  wrote:

> On 3/10/17, Sanchit Sinha  wrote:
> > Okay made some changes,
> >
>
> Do all changes, not just some.
> Also this must be full patch.
>
> You are missing Makefile, Changelog and so on...
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>



-- 
Sanchit Sinha
B.Tech- CSE
IIIT-Delhi
Roll-2015083
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Re: [FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Paul B Mahol]

On 3/10/17, Sanchit Sinha  wrote:
> Okay made some changes,
>

Do all changes, not just some.
Also this must be full patch.

You are missing Makefile, Changelog and so on...
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Re: [FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Sanchit Sinha]

Okay made some changes,

From c0c1a1e7d4ad1fcbd96827725a47af20145c7621 Mon Sep 17 00:00:00 2001
From: Sanchit Sinha 
Date: Sat, 11 Mar 2017 01:27:38 +0530
Subject: [PATCH] Changes to af_ambisonic.c, makefile and allfilters.c

---
 libavfilter/af_ambisonic.c | 136
+
 1 file changed, 136 insertions(+)
 create mode 100644 libavfilter/af_ambisonic.c

diff --git a/libavfilter/af_ambisonic.c b/libavfilter/af_ambisonic.c
new file mode 100644
index 000..03409a6
--- /dev/null
+++ b/libavfilter/af_ambisonic.c
@@ -0,0 +1,136 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
02110-1301 USA
+ */
+
+#include 
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include 
+
+typedef struct AmbisonicContext {
+ const AVClass *class;
+ /*Not needed, if any new variables are to be used, this struct can be
populated(f)*/
+
+} AmbisonicContext;
+
+#define OFFSET(x) offsetof(AmbisonicContext, x)
+
+static const AVOption ambisonic_options[] = {  //square will be an option
+{NULL}
+};
+
+AVFILTER_DEFINE_CLASS(ambisonic);
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layout = NULL;
+ int ret;
+ if ((ret = ff_add_format (, AV_SAMPLE_FMT_FLTP   )) < 0 ||
+ (ret = ff_set_common_formats (ctx, formats )) < 0 ||
+ (ret = ff_add_channel_layout ( , AV_CH_LAYOUT_4POINT0)) < 0 ||
+ (ret = ff_set_common_channel_layouts (ctx, layout   )) < 0)
+ return ret;
+ formats = ff_all_samplerates();
+ return ff_set_common_samplerates(ctx, formats);
+}
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ /*If variables used, this has to be created*/
+ // AmbisonicContext *s = ctx->priv;
+ // const float *src = (const float *)in->data[0];
+ // float *dst;
+ AVFrame *out;
+ int itr;
+ float *w,*x,*y;
+
+ if (av_frame_is_writable(in))
+ {
+ out = in;
+ }
+ else
+ {
+ out = ff_get_audio_buffer(inlink, in->nb_samples);
+ if (!out)
+ {
+ av_frame_free();
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
+
+ /*If planar samples are used, output must be put in dst*/
+ //dst = (float *)out->data[0];
+
+ float root8= sqrt(8);
+ float lf=0,lb=0,rb=0,rf=0;
+ for(itr=0;itrnb_samples;itr++)
+ {
+ /*Float pointers to the samples*/
+ w=(float *)(*(in->extended_data)+itr);
+ x=(float *)(*(in->extended_data+1)+itr);
+ y=(float *)(*(in->extended_data+2)+itr);
+
+ lf = root8 * (2*(*w)+*x+*y);
+ lb = root8 * (2*(*w)-*x+*y);
+ rb = root8 * (2*(*w)-*x-*y);
+ rf = root8 * (2*(*w)+*x-*y);
+
+ out->extended_data[0][itr]= lf;
+ out->extended_data[1][itr]= lb;
+ out->extended_data[2][itr]= rb;
+ out->extended_data[3][itr]= rf;
+ }
+
+ if (out != in)
+ av_frame_free();
+ return ff_filter_frame(outlink, out);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name= "default",
+ .type= AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ // .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_ambisonic = {
+ .name  = "ambisonic",
+ .description= NULL_IF_CONFIG_SMALL("An ambisonic filter"),
+ .query_formats  = query_formats,
+ .priv_size= sizeof(AmbisonicContext),
+ .priv_class  = _class,
+ // .uninit   = uninit,
+ .inputs  = inputs,
+ .outputs= outputs,
+};
\ No newline at end of file
-- 
2.7.4





On Sat, Mar 11, 2017 at 12:33 AM, Paul B Mahol  wrote:

> On 3/10/17, Sanchit Sinha  wrote:
> > libavfilter/af_ambisonic.c | 139
> > +
> >  1 file changed, 139 insertions(+)
> >  create mode 100644 libavfilter/af_ambisonic.c
> >
>
> Incomplete patch.
>
> > diff --git a/libavfilter/af_ambisonic.c b/libavfilter/af_ambisonic.c
> > new file mode 100644
> > index 000..98b0e44
> > --- /dev/null
> > +++ b/libavfilter/af_ambisonic.c
> > @@ -0,0 +1,139 @@
> > +/*
> > + * This file is part of FFmpeg.
> > + *
> > + * FFmpeg is free software; you can redistribute it and/or
> > + * 

Re: [FFmpeg-devel] [PATCH] avfilter/vf_interlace: add complex vertcal lowpassfilter

[James Almer]

On 3/8/2017 1:58 PM, Thomas Mundt wrote:
> Hi,
> 
> attached patch adds a complex (-1 2 6 2 -1) vertcal lowpassfilter to 
> vf_interlace. This will better retain detail and reduce blurring compared to 
> the existing (1 2 1) filter.
> 
> Please comment.
> 
> 
> 0001-avfilter-vf_interlace-add-complex-vertcal-lowpassfil.patch
> 
> 
> From f157bc9b898d1215f8ec10b301720a9d9ff03c63 Mon Sep 17 00:00:00 2001
> From: Thomas Mundt 
> Date: Wed, 8 Mar 2017 17:33:18 +0100
> Subject: [PATCH] avfilter/vf_interlace: add complex vertcal lowpassfilter Add
>  a complex (-1 2 6 2 -1) filter to reduce blurring compared to the existing (1
>  2 1) filter.
> 
> Signed-off-by: Thomas Mundt 
> ---
>  doc/filters.texi| 13 ++--
>  libavfilter/interlace.h |  2 ++
>  libavfilter/vf_interlace.c  | 42 ++--
>  libavfilter/x86/vf_interlace.asm| 65 
> +
>  libavfilter/x86/vf_interlace_init.c | 17 +++---
>  5 files changed, 130 insertions(+), 9 deletions(-)
> 
> diff --git a/doc/filters.texi b/doc/filters.texi
> index b5265d9..0041d39 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -9109,8 +9109,17 @@ This determines whether the interlaced frame is taken 
> from the even
>  (tff - default) or odd (bff) lines of the progressive frame.
>  
>  @item lowpass
> -Enable (default) or disable the vertical lowpass filter to avoid twitter
> -interlacing and reduce moire patterns.
> +Vertical lowpass filter to avoid twitter interlacing and
> +reduce moire patterns.
> +
> +@table @samp
> +@item 0
> +Disable vertical lowpass filter
> +@item 1
> +Enable linear filter (default)
> +@item 2
> +Enable complex filter
> +@end table
>  @end table
>  
>  @section kerndeint
> diff --git a/libavfilter/interlace.h b/libavfilter/interlace.h
> index da073ae..7ad457e 100644
> --- a/libavfilter/interlace.h
> +++ b/libavfilter/interlace.h
> @@ -51,6 +51,8 @@ typedef struct InterlaceContext {
>  AVFrame *cur, *next;   // the two frames from which the new one is 
> obtained
>  void (*lowpass_line)(uint8_t *dstp, ptrdiff_t linesize, const uint8_t 
> *srcp,
>   const uint8_t *srcp_above, const uint8_t 
> *srcp_below);
> +void (*lowpass_line_complex)(uint8_t *dstp, ptrdiff_t linesize,
> + const uint8_t *srcp, int mref, int pref);

Why not keep a single prototype, passing mref and pref for both linear
and complex? You can calculate srcp_above and srcp_below for linear like
you're doing it for complex in both the c and asm versions.

In any case, mref and pref should be ptrdiff_t and not int.

>  } InterlaceContext;
>  
>  void ff_interlace_init_x86(InterlaceContext *interlace);
> diff --git a/libavfilter/vf_interlace.c b/libavfilter/vf_interlace.c
> index efa3128..e8d5de4 100644
> --- a/libavfilter/vf_interlace.c
> +++ b/libavfilter/vf_interlace.c
> @@ -47,7 +47,7 @@ static const AVOption interlace_options[] = {
>  { "bff", "bottom field first", 0,
>  AV_OPT_TYPE_CONST, {.i64 = MODE_BFF }, INT_MIN, INT_MAX, .flags = 
> FLAGS, .unit = "scan" },
>  { "lowpass", "set vertical low-pass filter", OFFSET(lowpass),
> -AV_OPT_TYPE_BOOL,  {.i64 = 1 },0, 1, .flags = FLAGS },
> +AV_OPT_TYPE_INT,   {.i64 = 1 },0, 2, .flags = FLAGS },

Maybe add AV_OPT_TYPE_CONST values "off", "linear" and "complex".

>  { NULL }
>  };
>  
> @@ -67,6 +67,24 @@ static void lowpass_line_c(uint8_t *dstp, ptrdiff_t 
> linesize,
>  }
>  }
>  
> +static void lowpass_line_complex_c(uint8_t *dstp, ptrdiff_t linesize,
> +   const uint8_t *srcp, int mref, int pref)
> +{
> +const uint8_t *srcp_above = srcp + mref;
> +const uint8_t *srcp_below = srcp + pref;
> +const uint8_t *srcp_above2 = srcp + mref * 2;
> +const uint8_t *srcp_below2 = srcp + pref * 2;
> +int i;
> +for (i = 0; i < linesize; i++) {
> +// this calculation is an integer representation of
> +// '0.75 * current + 0.25 * above + 0.25 * below - 0.125 * above2 - 
> 0.125 * below2'
> +// '4 +' is for rounding.
> +dstp[i] = av_clip_uint8((4 + (srcp[i] << 2)
> +  + ((srcp[i] + srcp_above[i] + srcp_below[i]) << 1)
> +  - srcp_above2[i] - srcp_below2[i]) >> 3);
> +}
> +}
> +
>  static const enum AVPixelFormat formats_supported[] = {
>  AV_PIX_FMT_YUV420P,  AV_PIX_FMT_YUV422P,  AV_PIX_FMT_YUV444P,
>  AV_PIX_FMT_YUV444P,  AV_PIX_FMT_YUV410P,  AV_PIX_FMT_YUVA420P,
> @@ -116,7 +134,11 @@ static int config_out_props(AVFilterLink *outlink)
>  
>  
>  if (s->lowpass) {
> -s->lowpass_line = lowpass_line_c;
> +if (s->lowpass == 1)
> +s->lowpass_line = lowpass_line_c;
> +else if (s->lowpass == 2)
> +s->lowpass_line_complex = lowpass_line_complex_c;
> +
>  if (ARCH_X86)
>  

Re: [FFmpeg-devel] [PATCH] Add XPM decoder

[Lou Logan]

Hi,

On Sat, 11 Mar 2017 00:34:18 +0530
Paras  wrote:

> From: Paras Chadha 
> 
> Signed-off-by: Paras 
> ---
>  Changelog   |   1 +
>  doc/general.texi|   2 +
>  libavcodec/Makefile |   1 +
>  libavcodec/allcodecs.c  |   1 +
>  libavcodec/avcodec.h|   1 +
>  libavcodec/codec_desc.c |   7 +
>  libavcodec/version.h|   4 +-
>  libavcodec/xpmdec.c | 409 
> 

There are many instances of trailing whitespace in libavcodec/xpmdec.c.
This is to be avoided and prevents the patch from being applied.

If you use vim or emacs they can be configured to highlight trailing
whitespace (and also forbidden tabs):

http://ffmpeg.org/developer.html#Editor-configuration
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Re: [FFmpeg-devel] [PATCH] avcodec/vp8: Fix hang with slice threads

[Thomas Guilbert]

Thank you!

On Thu, Mar 9, 2017 at 5:20 PM, Michael Niedermayer 
wrote:

> On Thu, Mar 09, 2017 at 08:17:37PM -0500, Ronald S. Bultje wrote:
> > Hi,
> >
> > On Thu, Mar 9, 2017 at 8:12 PM, Michael Niedermayer
> 
> > wrote:
> >
> > > From: Thomas Guilbert 
> > >
> > > Fixes: 447860.webm
> > >
> > > Signed-off-by: Michael Niedermayer 
> > > ---
> > >  libavcodec/vp8.c | 4 +++-
> > >  1 file changed, 3 insertions(+), 1 deletion(-)
> > >
> > > diff --git a/libavcodec/vp8.c b/libavcodec/vp8.c
> > > index 6759b310f0..068223920e 100644
> > > --- a/libavcodec/vp8.c
> > > +++ b/libavcodec/vp8.c
> > > @@ -2508,8 +2508,10 @@ int vp78_decode_mb_row_sliced(AVCodecContext
> > > *avctx, void *tdata, int jobnr,
> > >  for (mb_y = jobnr; mb_y < s->mb_height; mb_y += num_jobs) {
> > >  td->thread_mb_pos = mb_y << 16;
> > >  ret = s->decode_mb_row_no_filter(avctx, tdata, jobnr,
> threadnr);
> > > -if (ret < 0)
> > > +if (ret < 0) {
> > > +update_pos(td, s->mb_height, INT_MAX & 0x);
> > >  return ret;
> > > +}
> > >  if (s->deblock_filter)
> > >  s->filter_mb_row(avctx, tdata, jobnr, threadnr);
> > >  update_pos(td, mb_y, INT_MAX & 0x);
> > > --
> > > 2.11.0
> >
> >
> > OK.
>
> applied
>
> thx
>
> [...]
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> Complexity theory is the science of finding the exact solution to an
> approximation. Benchmarking OTOH is finding an approximation of the exact
>
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[FFmpeg-devel] [PATCH] Add XPM decoder

[Paras]

From: Paras Chadha 

Signed-off-by: Paras 
---
 Changelog   |   1 +
 doc/general.texi|   2 +
 libavcodec/Makefile |   1 +
 libavcodec/allcodecs.c  |   1 +
 libavcodec/avcodec.h|   1 +
 libavcodec/codec_desc.c |   7 +
 libavcodec/version.h|   4 +-
 libavcodec/xpmdec.c | 409 
 libavformat/img2.c  |   1 +
 9 files changed, 425 insertions(+), 2 deletions(-)
 create mode 100644 libavcodec/xpmdec.c

diff --git a/Changelog b/Changelog
index 13628ca..716b6ff 100644
--- a/Changelog
+++ b/Changelog
@@ -26,6 +26,7 @@ version :
 - native Opus encoder
 - ScreenPressor decoder
 - incomplete ClearVideo decoder
+- XPM decoder
 
 version 3.2:
 - libopenmpt demuxer
diff --git a/doc/general.texi b/doc/general.texi
index 30450c0..83f54b3 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -607,6 +607,8 @@ following image formats are supported:
 @tab WebP image format, encoding supported through external library libwebp
 @item XBM  @tab X @tab X
 @tab X BitMap image format
+@item XPM  @tab X @tab X
+@tab X PixMap image format
 @item XFace @tab X @tab X
 @tab X-Face image format
 @item XWD  @tab X @tab X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 65ccbad..fc5fa1f 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -645,6 +645,7 @@ OBJS-$(CONFIG_XAN_WC4_DECODER) += xxan.o
 OBJS-$(CONFIG_XBIN_DECODER)+= bintext.o cga_data.o
 OBJS-$(CONFIG_XBM_DECODER) += xbmdec.o
 OBJS-$(CONFIG_XBM_ENCODER) += xbmenc.o
+OBJS-$(CONFIG_XPM_DECODER) += xpmdec.o
 OBJS-$(CONFIG_XFACE_DECODER)   += xfacedec.o xface.o
 OBJS-$(CONFIG_XFACE_ENCODER)   += xfaceenc.o xface.o
 OBJS-$(CONFIG_XL_DECODER)  += xl.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 074efd4..81208b2 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -376,6 +376,7 @@ static void register_all(void)
 REGISTER_DECODER(XAN_WC3,   xan_wc3);
 REGISTER_DECODER(XAN_WC4,   xan_wc4);
 REGISTER_ENCDEC (XBM,   xbm);
+REGISTER_DECODER (XPM,   xpm);
 REGISTER_ENCDEC (XFACE, xface);
 REGISTER_DECODER(XL,xl);
 REGISTER_ENCDEC (XWD,   xwd);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 30ac236..a862716 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -377,6 +377,7 @@ enum AVCodecID {
 AV_CODEC_ID_XWD,
 AV_CODEC_ID_CDXL,
 AV_CODEC_ID_XBM,
+AV_CODEC_ID_XPM,
 AV_CODEC_ID_ZEROCODEC,
 AV_CODEC_ID_MSS1,
 AV_CODEC_ID_MSA1,
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index 06bcfc3..88cfddb 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -1591,6 +1591,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
 .props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSLESS,
 },
 {
+.id= AV_CODEC_ID_XPM,
+.type  = AVMEDIA_TYPE_VIDEO,
+.name  = "xpm",
+.long_name = NULL_IF_CONFIG_SMALL("XPM (X PixMap) image"),
+.props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSLESS,
+},
+{
 .id= AV_CODEC_ID_XWD,
 .type  = AVMEDIA_TYPE_VIDEO,
 .name  = "xwd",
diff --git a/libavcodec/version.h b/libavcodec/version.h
index b00e011..3ed5a71 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -28,8 +28,8 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR  57
-#define LIBAVCODEC_VERSION_MINOR  82
-#define LIBAVCODEC_VERSION_MICRO 102
+#define LIBAVCODEC_VERSION_MINOR  83
+#define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
diff --git a/libavcodec/xpmdec.c b/libavcodec/xpmdec.c
new file mode 100644
index 000..635f25a
--- /dev/null
+++ b/libavcodec/xpmdec.c
@@ -0,0 +1,409 @@
+/*
+ * XPM image format
+ *
+ * Copyright (c) 2012 Paul B Mahol
+ * Copyright (c) 2016 Paras Chadha
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 

Re: [FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Paul B Mahol]

On 3/10/17, Sanchit Sinha  wrote:
> libavfilter/af_ambisonic.c | 139
> +
>  1 file changed, 139 insertions(+)
>  create mode 100644 libavfilter/af_ambisonic.c
>

Incomplete patch.

> diff --git a/libavfilter/af_ambisonic.c b/libavfilter/af_ambisonic.c
> new file mode 100644
> index 000..98b0e44
> --- /dev/null
> +++ b/libavfilter/af_ambisonic.c
> @@ -0,0 +1,139 @@
> +/*
> + * This file is part of FFmpeg.
> + *
> + * FFmpeg is free software; you can redistribute it and/or
> + * modify it under the terms of the GNU Lesser General Public
> + * License as published by the Free Software Foundation; either
> + * version 2.1 of the License, or (at your option) any later version.
> + *
> + * FFmpeg is distributed in the hope that it will be useful,
> + * but WITHOUT ANY WARRANTY; without even the implied warranty of
> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
> + * Lesser General Public License for more details.
> + *
> + * You should have received a copy of the GNU Lesser General Public
> + * License along with FFmpeg; if not, write to the Free Software
> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
> 02110-1301 USA
> + */
> +
> +#include 
> +#include "libavutil/avstring.h"
> +#include "libavutil/channel_layout.h"
> +#include "libavutil/opt.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +
> +#define root8 2.828

One can use sqrt(8) just fine.

> +
> +typedef struct AmbisonicContext {
> +const AVClass *class;
> +/*Not needed, if any new variables are to be used, this struct can be
> populated*/
> +
> +} AmbisonicContext;
> +
> +#define OFFSET(x) offsetof(AmbisonicContext, x) //Future use(maybe)
> +
> +static const AVOption ambisonic_options[] = {  //square will be an option
> +{NULL}
> +};
> +
> +AVFILTER_DEFINE_CLASS(ambisonic);
> +static int query_formats(AVFilterContext *ctx)
> +{
> +AVFilterFormats *formats = NULL;
> +AVFilterChannelLayouts *layout = NULL;
> +int ret;
> +if ((ret = ff_add_format (, AV_SAMPLE_FMT_FLTP
>   )) < 0 ||
> +(ret = ff_set_common_formats (ctx , formats
>  )) < 0 ||
> +(ret = ff_add_channel_layout ( ,
> AV_CH_LAYOUT_4POINT0)) < 0 ||
> +(ret = ff_set_common_channel_layouts (ctx , layout
> )) < 0)
> +return ret;
> +formats = ff_all_samplerates();
> +return ff_set_common_samplerates(ctx, formats);
> +}
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> +AVFilterContext *ctx = inlink->dst;
> +AVFilterLink *outlink = ctx->outputs[0];
> +/*If variables used, this has to be created*/
> +// AmbisonicContext *s = ctx->priv;
> +// const float *src = (const float *)in->data[0];
> +// float *dst;
> +float *lf,*lb,*rb,*rf;
> +AVFrame *out;
> +int itr;
> +float *w,*x,*y;
> +
> +if (av_frame_is_writable(in)) {
> +out = in;
> +} else {
> +out = ff_get_audio_buffer(inlink, in->nb_samples);
> +if (!out) {
> +av_frame_free();
> +return AVERROR(ENOMEM);
> +}
> +av_frame_copy_props(out, in);
> +}
> +
> +/*If planar samples are used, output must be put in dst*/
> +//dst = (float *)out->data[0];
> +
> +lf = (float *)malloc(sizeof(float));
> +lb = (float *)malloc(sizeof(float));
> +rb = (float *)malloc(sizeof(float));
> +rf = (float *)malloc(sizeof(float));
> +

Why? This is unacceptable. Use normal float variables.

> +for(itr=0;itrnb_samples;itr++)
> +{
> +*lf=0,*lb=0,*rb=0,*rf=0;
> +/*Float pointers to the samples*/
> +w=(float *)(*(in->extended_data)+itr);
> +x=(float *)(*(in->extended_data+1)+itr);
> +y=(float *)(*(in->extended_data+2)+itr);

This can be simplified and moved above loop.
Good understanding of pointers is mandatory.

> +
> +*lf = root8 * (2*(*w)+*x+*y);
> +*lb = root8 * (2*(*w)-*x+*y);
> +*rb = root8 * (2*(*w)-*x-*y);
> +*rf = root8 * (2*(*w)+*x-*y);
> +
> +/*Mathematical coefficients taken from :
> https://en.wikipedia.org/wiki/Ambisonics*/

Remove this comment.

> +out->extended_data[0][itr]= *lf;
> +out->extended_data[1][itr]= *lb;
> +out->extended_data[2][itr]= *rb;
> +out->extended_data[3][itr]= *rf;
> +}
> +
> +if (out != in)
> +av_frame_free();
> +return ff_filter_frame(outlink, out);
> +}
> +
> +static const AVFilterPad inputs[] = {
> +{
> +.name = "default",
> +.type = AVMEDIA_TYPE_AUDIO,
> +.filter_frame = filter_frame,
> +// .config_props = config_input,
> +},
> +{ NULL }
> +};
> +
> +static const AVFilterPad outputs[] = {
> +{
> +.name = "default",
> +.type = AVMEDIA_TYPE_AUDIO,
> +},
> +{ NULL }
> +};
> +
> +AVFilter ff_af_ambisonic = {
> +

Re: [FFmpeg-devel] [PATCH] avfilter/vf_interlace: add complex vertcal lowpassfilter

[Kieran Kunhya]

Hi,

Just wondering, did you see what this looks like on a CRT, especially with
interline twitter.
The original method is similar to what an interlaced camera does, averaging
alternate lines, deliberately lowpassing the data.

Regards,
Kieran Kunhya

On Wed, 8 Mar 2017 at 16:58 Thomas Mundt 
wrote:

> Hi,
>
> attached patch adds a complex (-1 2 6 2 -1) vertcal lowpassfilter to
> vf_interlace. This will better retain detail and reduce blurring compared
> to the existing (1 2 1) filter.
>
> Please comment.___
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
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[FFmpeg-devel] [PATCH] libavfilter/af_ambisonic.c Added File for Ambisonic Filter

[Sanchit Sinha]

libavfilter/af_ambisonic.c | 139
+
 1 file changed, 139 insertions(+)
 create mode 100644 libavfilter/af_ambisonic.c

diff --git a/libavfilter/af_ambisonic.c b/libavfilter/af_ambisonic.c
new file mode 100644
index 000..98b0e44
--- /dev/null
+++ b/libavfilter/af_ambisonic.c
@@ -0,0 +1,139 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
02110-1301 USA
+ */
+
+#include 
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+
+#define root8 2.828
+
+typedef struct AmbisonicContext {
+const AVClass *class;
+/*Not needed, if any new variables are to be used, this struct can be
populated*/
+
+} AmbisonicContext;
+
+#define OFFSET(x) offsetof(AmbisonicContext, x) //Future use(maybe)
+
+static const AVOption ambisonic_options[] = {  //square will be an option
+{NULL}
+};
+
+AVFILTER_DEFINE_CLASS(ambisonic);
+static int query_formats(AVFilterContext *ctx)
+{
+AVFilterFormats *formats = NULL;
+AVFilterChannelLayouts *layout = NULL;
+int ret;
+if ((ret = ff_add_format (, AV_SAMPLE_FMT_FLTP
  )) < 0 ||
+(ret = ff_set_common_formats (ctx , formats
 )) < 0 ||
+(ret = ff_add_channel_layout ( ,
AV_CH_LAYOUT_4POINT0)) < 0 ||
+(ret = ff_set_common_channel_layouts (ctx , layout
)) < 0)
+return ret;
+formats = ff_all_samplerates();
+return ff_set_common_samplerates(ctx, formats);
+}
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+AVFilterContext *ctx = inlink->dst;
+AVFilterLink *outlink = ctx->outputs[0];
+/*If variables used, this has to be created*/
+// AmbisonicContext *s = ctx->priv;
+// const float *src = (const float *)in->data[0];
+// float *dst;
+float *lf,*lb,*rb,*rf;
+AVFrame *out;
+int itr;
+float *w,*x,*y;
+
+if (av_frame_is_writable(in)) {
+out = in;
+} else {
+out = ff_get_audio_buffer(inlink, in->nb_samples);
+if (!out) {
+av_frame_free();
+return AVERROR(ENOMEM);
+}
+av_frame_copy_props(out, in);
+}
+
+/*If planar samples are used, output must be put in dst*/
+//dst = (float *)out->data[0];
+
+lf = (float *)malloc(sizeof(float));
+lb = (float *)malloc(sizeof(float));
+rb = (float *)malloc(sizeof(float));
+rf = (float *)malloc(sizeof(float));
+
+for(itr=0;itrnb_samples;itr++)
+{
+*lf=0,*lb=0,*rb=0,*rf=0;
+/*Float pointers to the samples*/
+w=(float *)(*(in->extended_data)+itr);
+x=(float *)(*(in->extended_data+1)+itr);
+y=(float *)(*(in->extended_data+2)+itr);
+
+*lf = root8 * (2*(*w)+*x+*y);
+*lb = root8 * (2*(*w)-*x+*y);
+*rb = root8 * (2*(*w)-*x-*y);
+*rf = root8 * (2*(*w)+*x-*y);
+
+/*Mathematical coefficients taken from :
https://en.wikipedia.org/wiki/Ambisonics*/
+out->extended_data[0][itr]= *lf;
+out->extended_data[1][itr]= *lb;
+out->extended_data[2][itr]= *rb;
+out->extended_data[3][itr]= *rf;
+}
+
+if (out != in)
+av_frame_free();
+return ff_filter_frame(outlink, out);
+}
+
+static const AVFilterPad inputs[] = {
+{
+.name = "default",
+.type = AVMEDIA_TYPE_AUDIO,
+.filter_frame = filter_frame,
+// .config_props = config_input,
+},
+{ NULL }
+};
+
+static const AVFilterPad outputs[] = {
+{
+.name = "default",
+.type = AVMEDIA_TYPE_AUDIO,
+},
+{ NULL }
+};
+
+AVFilter ff_af_ambisonic = {
+.name   = "ambisonic",
+.description= NULL_IF_CONFIG_SMALL("An ambisonic filter"),
+.query_formats  = query_formats,
+.priv_size  = sizeof(AmbisonicContext),
+.priv_class = _class,
+// .uninit = uninit,
+.inputs = inputs,
+.outputs= outputs,
+};
\ No newline at end of file
-- 
2.7.4
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Re: [FFmpeg-devel] [PATCH 1/2] avcodec/dca_xll: Fix runtime error: signed integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'

[Hendrik Leppkes]

On Fri, Mar 10, 2017 at 3:24 PM, Michael Niedermayer
 wrote:
> Fixes: 732/clusterfuzz-testcase-4872990070145024
>
> See: [FFmpeg-devel] [PATCH 2/6] avcodec/dca_xll: Fix runtime error: signed 
> integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'
> Found-by: continuous fuzzing process 
> https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
> Signed-off-by: Michael Niedermayer 
> ---
>  libavcodec/dca_xll.c | 2 +-
>  1 file changed, 1 insertion(+), 1 deletion(-)
>
> diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c
> index 6cebda35e4..0fe90c7348 100644
> --- a/libavcodec/dca_xll.c
> +++ b/libavcodec/dca_xll.c
> @@ -1312,7 +1312,7 @@ static int combine_residual_frame(DCAXllDecoder *s, 
> DCAXllChSet *c)
>  } else {
>  // No downmix scaling
>  for (n = 0; n < nsamples; n++)
> -dst[n] += (src[n] + round) >> shift;
> +dst[n] += (unsigned)((src[n] + round) >> shift);
>  }
>  }
>

audio samples are typically signed, using negative values too, is this
really safe to just cast it like that, without modifying the value in
the end?

- Hendrik
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Re: [FFmpeg-devel] configure: clang -Oz for small size build to reduce size further

[James Almer]

On 3/10/2017 12:43 PM, Paul B Mahol wrote:
> On 3/10/17, James Almer  wrote:
>> On 3/10/2017 10:50 AM, Paul B Mahol wrote:
>>> On 3/10/17, wm4  wrote:
 On Fri, 10 Mar 2017 13:47:38 +0100
 Paul B Mahol  wrote:

> On 3/10/17, Wang Bin  wrote:
>>
>>
>
> I don't think this is good idea.

 Why not?
>>>
>>> Because it changes something that was working before.
>>
>> The only valid reason i could see to not go with this change is
>> if the size gains are minimal and the speed loss considerable.
>> Otherwise, if the end result is smaller than -Os then there's no
>> reason to not use it.
> 
> OP didn't posted results at all, so applying this would be blind move
> IMHO.

Agree. Some numbers would be good (resulting binary size, benchmarks
of at least popular codecs, etc).

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Re: [FFmpeg-devel] configure: clang -Oz for small size build to reduce size further

[Paul B Mahol]

On 3/10/17, James Almer  wrote:
> On 3/10/2017 10:50 AM, Paul B Mahol wrote:
>> On 3/10/17, wm4  wrote:
>>> On Fri, 10 Mar 2017 13:47:38 +0100
>>> Paul B Mahol  wrote:
>>>
 On 3/10/17, Wang Bin  wrote:
>
>

 I don't think this is good idea.
>>>
>>> Why not?
>>
>> Because it changes something that was working before.
>
> The only valid reason i could see to not go with this change is
> if the size gains are minimal and the speed loss considerable.
> Otherwise, if the end result is smaller than -Os then there's no
> reason to not use it.

OP didn't posted results at all, so applying this would be blind move
IMHO.
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Re: [FFmpeg-devel] [PATCH] lavf/dashenc: update bitrates on dash_write_trailer

[Przemysław Sobala]

On Wed, Mar 8, 2017 at 9:27 AM, Przemysław Sobala <
przemyslaw.sob...@gmail.com> wrote:

> On Fri, Mar 3, 2017 at 9:38 AM, Przemysław Sobala <
> przemyslaw.sob...@gmail.com> wrote:
>
>> From: Przemysław Sobala 
>>
>> Provides a way to change bandwidth parameter inside DASH manifest after a
>> non-CBR H.264 encoding.
>> Caller now is able to compute the bitrate by itself, after all packets
>> have been written, and then set that value in 
>> AVFormatContext->streams->codecpar->bit_rate
>> before calling av_write_trailer. As a result that value will be set in DASH
>> manifest.
>> ---
>>  libavformat/dashenc.c | 42 ++
>>  1 file changed, 30 insertions(+), 12 deletions(-)
>>
>> diff --git a/libavformat/dashenc.c b/libavformat/dashenc.c
>> index fa56505..011d2ea 100644
>> --- a/libavformat/dashenc.c
>> +++ b/libavformat/dashenc.c
>> @@ -561,6 +561,30 @@ static int write_manifest(AVFormatContext *s, int
>> final)
>>  return 0;
>>  }
>>
>> +static int set_bitrate(AVFormatContext *s)
>> +{
>> +DASHContext *c = s->priv_data;
>> +int i;
>> +
>> +for (i = 0; i < s->nb_streams; i++) {
>> +OutputStream *os = >streams[i];
>> +
>> +os->bit_rate = s->streams[i]->codecpar->bit_rate;
>> +if (os->bit_rate) {
>> +snprintf(os->bandwidth_str, sizeof(os->bandwidth_str),
>> + " bandwidth=\"%d\"", os->bit_rate);
>> +} else {
>> +int level = s->strict_std_compliance >= FF_COMPLIANCE_STRICT
>> ?
>> +AV_LOG_ERROR : AV_LOG_WARNING;
>> +av_log(s, level, "No bit rate set for stream %d\n", i);
>> +if (s->strict_std_compliance >= FF_COMPLIANCE_STRICT)
>> +return AVERROR(EINVAL);
>> +}
>> +}
>> +
>> +return 0;
>> +}
>> +
>>  static int dash_init(AVFormatContext *s)
>>  {
>>  DASHContext *c = s->priv_data;
>> @@ -597,6 +621,10 @@ static int dash_init(AVFormatContext *s)
>>  if (!c->streams)
>>  return AVERROR(ENOMEM);
>>
>> +ret = set_bitrate(s);
>> +if (ret < 0)
>> +return ret;
>> +
>>  for (i = 0; i < s->nb_streams; i++) {
>>  OutputStream *os = >streams[i];
>>  AVFormatContext *ctx;
>> @@ -604,18 +632,6 @@ static int dash_init(AVFormatContext *s)
>>  AVDictionary *opts = NULL;
>>  char filename[1024];
>>
>> -os->bit_rate = s->streams[i]->codecpar->bit_rate;
>> -if (os->bit_rate) {
>> -snprintf(os->bandwidth_str, sizeof(os->bandwidth_str),
>> - " bandwidth=\"%d\"", os->bit_rate);
>> -} else {
>> -int level = s->strict_std_compliance >= FF_COMPLIANCE_STRICT
>> ?
>> -AV_LOG_ERROR : AV_LOG_WARNING;
>> -av_log(s, level, "No bit rate set for stream %d\n", i);
>> -if (s->strict_std_compliance >= FF_COMPLIANCE_STRICT)
>> -return AVERROR(EINVAL);
>> -}
>> -
>>  ctx = avformat_alloc_context();
>>  if (!ctx)
>>  return AVERROR(ENOMEM);
>> @@ -981,6 +997,8 @@ static int dash_write_trailer(AVFormatContext *s)
>>  {
>>  DASHContext *c = s->priv_data;
>>
>> +set_bitrate(s);
>> +
>>  if (s->nb_streams > 0) {
>>  OutputStream *os = >streams[0];
>>  // If no segments have been written so far, try to do a crude
>> --
>> 2.7.4
>>
>>
> ping
>
>
ping, 7 days timeout
can anyone tak a look at this?
[
http://ffmpeg.org/developer.html#Always-wait-long-enough-before-pushing-changes
]

--
Regards
Przemysław Sobala
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Re: [FFmpeg-devel] [PATCH 2/2] avcodec/h264_direct: Fix runtime error: signed integer overflow: 2147483647 - -14133 cannot be represented in type 'int'

[wm4]

On Fri, 10 Mar 2017 15:24:52 +0100
Michael Niedermayer  wrote:

> Fixes: 755/clusterfuzz-testcase-5369072516595712
> 
> See: [FFmpeg-devel] [PATCH 1/2] avcodec/h264_direct: Fix runtime error: 
> signed integer overflow: 2147483647 - -14133 cannot be represented in type 
> 'int'
> 
> Found-by: continuous fuzzing process 
> https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
> Signed-off-by: Michael Niedermayer 
> ---
>  libavcodec/h264_direct.c | 7 ++-
>  1 file changed, 6 insertions(+), 1 deletion(-)
> 
> diff --git a/libavcodec/h264_direct.c b/libavcodec/h264_direct.c
> index cbb84665b3..66e54479d1 100644
> --- a/libavcodec/h264_direct.c
> +++ b/libavcodec/h264_direct.c
> @@ -39,7 +39,12 @@ static int get_scale_factor(H264SliceContext *sl,
>  int poc, int poc1, int i)
>  {
>  int poc0 = sl->ref_list[0][i].poc;
> -int td = av_clip_int8(poc1 - poc0);
> +int64_t pocdiff = poc1 - (int64_t)poc0;
> +int td = av_clip_int8(pocdiff);
> +
> +if (pocdiff != (int)pocdiff)
> +avpriv_request_sample(sl->h264->avctx, "pocdiff overflow\n");
> +
>  if (td == 0 || sl->ref_list[0][i].parent->long_ref) {
>  return 256;
>  } else {

Hard to image that these poc values aren't bounded by something else,
but I don't know.

Also the previous patch didn't have this request_sample call, which
inflates this whole thing by 5 lines of code.
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Re: [FFmpeg-devel] [PATCH] swresample/options: enable linear_interp and exact_rational by default

[Muhammad Faiz]

On Fri, Mar 10, 2017 at 5:44 PM, Michael Niedermayer
 wrote:
> On Fri, Mar 10, 2017 at 11:11:05AM +0700, Muhammad Faiz wrote:
>> On Fri, Mar 10, 2017 at 10:44 AM, Michael Niedermayer
>>  wrote:
>> > On Fri, Mar 10, 2017 at 09:19:38AM +0700, Muhammad Faiz wrote:
>> >> for higher quality resampling
>> >
>> > can you hear the difference ?
>>
>> Yes, but it is sinesweep test audio, not real music audio.
>
> my question was meant to be, can you hear the difference on real
> world audio ?

No, I can not. I'm untrained for that. Because real world audio is complex,
it is difficult for me to judge the difference. Other people may hear it.

is sinesweep test not enough for this?
exact_rational won't hurt speed at all.

Thank's
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Re: [FFmpeg-devel] [PATCH 1/2] avcodec/dca_xll: Fix runtime error: signed integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'

[wm4]

On Fri, 10 Mar 2017 15:24:51 +0100
Michael Niedermayer  wrote:

> Fixes: 732/clusterfuzz-testcase-4872990070145024
> 
> See: [FFmpeg-devel] [PATCH 2/6] avcodec/dca_xll: Fix runtime error: signed 
> integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'
> Found-by: continuous fuzzing process 
> https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
> Signed-off-by: Michael Niedermayer 
> ---
>  libavcodec/dca_xll.c | 2 +-
>  1 file changed, 1 insertion(+), 1 deletion(-)
> 
> diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c
> index 6cebda35e4..0fe90c7348 100644
> --- a/libavcodec/dca_xll.c
> +++ b/libavcodec/dca_xll.c
> @@ -1312,7 +1312,7 @@ static int combine_residual_frame(DCAXllDecoder *s, 
> DCAXllChSet *c)
>  } else {
>  // No downmix scaling
>  for (n = 0; n < nsamples; n++)
> -dst[n] += (src[n] + round) >> shift;
> +dst[n] += (unsigned)((src[n] + round) >> shift);
>  }
>  }
>  

Seems reasonable.
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Re: [FFmpeg-devel] [PATCH 3/3] ffmpeg, ffprobe: don't "merge" side data into packet data by default

[wm4]

On Fri, 10 Mar 2017 15:29:17 +0100
Michael Niedermayer  wrote:

> Hi
> 
> On Thu, Mar 09, 2017 at 03:45:19PM +0100, wm4 wrote:
> > Preparation for potentially disabling merged side data by default in the
> > libs. Do this in particular because it affects fate tests.
> > 
> > The changed tests either reflect added packet side data, or the changed
> > packet size due to merged side data removal reducing the packet size.
> > 
> > The previous patches are needed because there is apparently no other way
> > for ffmpeg.c to pass the flag down to sub demuxers without potentially
> > overriding command line settings.
> > ---
> >  ffmpeg_opt.c  |   2 +
> >  ffprobe.c |   8 ++
> >  libavformat/tests/seek.c  |   2 +
> >  tests/ref/fate/concat-demuxer-simple2-lavf-ts | 192 
> > +-
> >  tests/ref/fate/gaplessenc-itunes-to-ipod-aac  |   4 +-
> >  tests/ref/fate/gaplessenc-pcm-to-mov-aac  |   4 +-
> >  tests/ref/fate/gaplessinfo-itunes1|   4 +-
> >  tests/ref/fate/gaplessinfo-itunes2|   4 +-
> >  tests/ref/fate/mov-aac-2048-priming   |   3 +-
> >  tests/ref/seek/cache-pipe |  10 +-
> >  tests/ref/seek/extra-mp3  |  18 +--
> >  tests/ref/seek/lavf-ts|  54 
> >  tests/ref/seek/mkv-codec-delay|  14 +-
> >  13 files changed, 198 insertions(+), 121 deletions(-)  
> 
> just wanted to say that iam not sure i will have time to test this
> in the near future, thus others should review / approve
> 
> saying so theres no misunderstanding if i dont reply with an approval
> or tested-by 
> 
> sorry & thx

OK, then I'll push on Monday.
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[FFmpeg-devel] [PATCH 2/2] avcodec/h264_direct: Fix runtime error: signed integer overflow: 2147483647 - -14133 cannot be represented in type 'int'

[Michael Niedermayer]

Fixes: 755/clusterfuzz-testcase-5369072516595712

See: [FFmpeg-devel] [PATCH 1/2] avcodec/h264_direct: Fix runtime error: signed 
integer overflow: 2147483647 - -14133 cannot be represented in type 'int'

Found-by: continuous fuzzing process 
https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer 
---
 libavcodec/h264_direct.c | 7 ++-
 1 file changed, 6 insertions(+), 1 deletion(-)

diff --git a/libavcodec/h264_direct.c b/libavcodec/h264_direct.c
index cbb84665b3..66e54479d1 100644
--- a/libavcodec/h264_direct.c
+++ b/libavcodec/h264_direct.c
@@ -39,7 +39,12 @@ static int get_scale_factor(H264SliceContext *sl,
 int poc, int poc1, int i)
 {
 int poc0 = sl->ref_list[0][i].poc;
-int td = av_clip_int8(poc1 - poc0);
+int64_t pocdiff = poc1 - (int64_t)poc0;
+int td = av_clip_int8(pocdiff);
+
+if (pocdiff != (int)pocdiff)
+avpriv_request_sample(sl->h264->avctx, "pocdiff overflow\n");
+
 if (td == 0 || sl->ref_list[0][i].parent->long_ref) {
 return 256;
 } else {
-- 
2.11.0

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[FFmpeg-devel] [PATCH 1/2] avcodec/dca_xll: Fix runtime error: signed integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'

[Michael Niedermayer]

Fixes: 732/clusterfuzz-testcase-4872990070145024

See: [FFmpeg-devel] [PATCH 2/6] avcodec/dca_xll: Fix runtime error: signed 
integer overflow: 2147286116 + 6298923 cannot be represented in type 'int'
Found-by: continuous fuzzing process 
https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer 
---
 libavcodec/dca_xll.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c
index 6cebda35e4..0fe90c7348 100644
--- a/libavcodec/dca_xll.c
+++ b/libavcodec/dca_xll.c
@@ -1312,7 +1312,7 @@ static int combine_residual_frame(DCAXllDecoder *s, 
DCAXllChSet *c)
 } else {
 // No downmix scaling
 for (n = 0; n < nsamples; n++)
-dst[n] += (src[n] + round) >> shift;
+dst[n] += (unsigned)((src[n] + round) >> shift);
 }
 }
 
-- 
2.11.0

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Re: [FFmpeg-devel] [PATCH 3/3] ffmpeg, ffprobe: don't "merge" side data into packet data by default

[Michael Niedermayer]

Hi

On Thu, Mar 09, 2017 at 03:45:19PM +0100, wm4 wrote:
> Preparation for potentially disabling merged side data by default in the
> libs. Do this in particular because it affects fate tests.
> 
> The changed tests either reflect added packet side data, or the changed
> packet size due to merged side data removal reducing the packet size.
> 
> The previous patches are needed because there is apparently no other way
> for ffmpeg.c to pass the flag down to sub demuxers without potentially
> overriding command line settings.
> ---
>  ffmpeg_opt.c  |   2 +
>  ffprobe.c |   8 ++
>  libavformat/tests/seek.c  |   2 +
>  tests/ref/fate/concat-demuxer-simple2-lavf-ts | 192 
> +-
>  tests/ref/fate/gaplessenc-itunes-to-ipod-aac  |   4 +-
>  tests/ref/fate/gaplessenc-pcm-to-mov-aac  |   4 +-
>  tests/ref/fate/gaplessinfo-itunes1|   4 +-
>  tests/ref/fate/gaplessinfo-itunes2|   4 +-
>  tests/ref/fate/mov-aac-2048-priming   |   3 +-
>  tests/ref/seek/cache-pipe |  10 +-
>  tests/ref/seek/extra-mp3  |  18 +--
>  tests/ref/seek/lavf-ts|  54 
>  tests/ref/seek/mkv-codec-delay|  14 +-
>  13 files changed, 198 insertions(+), 121 deletions(-)

just wanted to say that iam not sure i will have time to test this
in the near future, thus others should review / approve

saying so theres no misunderstanding if i dont reply with an approval
or tested-by 

sorry & thx

[...]
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Re: [FFmpeg-devel] configure: clang -Oz for small size build to reduce size further

[James Almer]

On 3/10/2017 10:50 AM, Paul B Mahol wrote:
> On 3/10/17, wm4  wrote:
>> On Fri, 10 Mar 2017 13:47:38 +0100
>> Paul B Mahol  wrote:
>>
>>> On 3/10/17, Wang Bin  wrote:


>>>
>>> I don't think this is good idea.
>>
>> Why not?
> 
> Because it changes something that was working before.

The only valid reason i could see to not go with this change is
if the size gains are minimal and the speed loss considerable.
Otherwise, if the end result is smaller than -Os then there's no
reason to not use it.

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Re: [FFmpeg-devel] configure: clang -Oz for small size build to reduce size further

[wm4]

On Fri, 10 Mar 2017 14:50:32 +0100
Paul B Mahol  wrote:

> On 3/10/17, wm4  wrote:
> > On Fri, 10 Mar 2017 13:47:38 +0100
> > Paul B Mahol  wrote:
> >  
> >> On 3/10/17, Wang Bin  wrote:  
> >> >
> >> >  
> >>
> >> I don't think this is good idea.  
> >
> > Why not?  
> 
> Because it changes something that was working before.

That's not a good reason at all.
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Re: [FFmpeg-devel] configure: clang -Oz for small size build to reduce size further

[Paul B Mahol]

On 3/10/17, wm4  wrote:
> On Fri, 10 Mar 2017 13:47:38 +0100
> Paul B Mahol  wrote:
>
>> On 3/10/17, Wang Bin  wrote:
>> >
>> >
>>
>> I don't think this is good idea.
>
> Why not?

Because it changes something that was working before.
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Re: [FFmpeg-devel] [PATCH] libavfilter/vf_detelecine: Added documentation to clarify the workings of the filter.

[Carl Eugen Hoyos]

2017-03-09 9:20 GMT+01:00  :

> I hope this patch adding code documentation to the detelecine filter
> will help with maintenance.

If you are interested in working on detelecine, please look at ticket #5662:
It is currently not possible to choose (with an option) which of two
subsequent duplicated frames are dropped / for the sample there,
the progressive (complete) frame is dropped, instead a composed
frame is dropped: They should be identical, but because of reencoding
the composed frame looks bad, I believe we cannot rule out that this
can also happen for distributed material.

Carl Eugen
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Re: [FFmpeg-devel] configure: clang -Oz for small size build to reduce size further

[wm4]

On Fri, 10 Mar 2017 13:47:38 +0100
Paul B Mahol  wrote:

> On 3/10/17, Wang Bin  wrote:
> >
> >  
> 
> I don't think this is good idea.

Why not?
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Re: [FFmpeg-devel] [PATCH] libavcodec/libopenjpegenc.c

[Aaron Boxer]

On Fri, Mar 10, 2017 at 12:00 AM, Michael Bradshaw 
wrote:

> On Thu, Mar 9, 2017 at 8:09 PM, Aaron Boxer  wrote:
> >
> > Thanks, Michael. Here are those changes.
>
>
> Applied. Thanks for the patch!
>


My pleasure. If there are any changes people are interested in for openjpeg
encode/decode, let me know and I would be happy to work on it.




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Re: [FFmpeg-devel] [PATCH] libavcodec/libopenjpegenc.c

[Aaron Boxer]

On Fri, Mar 10, 2017 at 7:52 AM, Paul B Mahol  wrote:

> On 3/10/17, Michael Bradshaw  wrote:
> > On Thu, Mar 9, 2017 at 8:09 PM, Aaron Boxer  wrote:
> >>
> >> Thanks, Michael. Here are those changes.
> >
> >
> > Applied. Thanks for the patch!
>
> AFAIK some of those could make use of BOOL.
>

Yes, good point.  Although, the OpenJPEG parameters are not BOOL, and since
the AVOptions
are mostly mirrors of OpenJPEG parameters, it might make sense to leave
them as they are.





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Re: [FFmpeg-devel] [PATCH] swresample/options: enable linear_interp and exact_rational by default

[Michael Niedermayer]

On Fri, Mar 10, 2017 at 01:49:41PM +0100, Paul B Mahol wrote:
> On 3/10/17, Michael Niedermayer  wrote:
> > On Fri, Mar 10, 2017 at 11:11:05AM +0700, Muhammad Faiz wrote:
> >> On Fri, Mar 10, 2017 at 10:44 AM, Michael Niedermayer
> >>  wrote:
> >> > On Fri, Mar 10, 2017 at 09:19:38AM +0700, Muhammad Faiz wrote:
> >> >> for higher quality resampling
> >> >
> >> > can you hear the difference ?
> >>
> >> Yes, but it is sinesweep test audio, not real music audio.
> >
> > my question was meant to be, can you hear the difference on real
> > world audio ?
> >
> 
> Yes I can hear it. I made blind test.

iam surprised, i didnt expect this to be audible with real world audio
what audio/music is this audible with ?

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Re: [FFmpeg-devel] configure: clang -Oz for small size build to reduce size further

[Paul B Mahol]

On 3/10/17, Wang Bin  wrote:
>
>

I don't think this is good idea.
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Re: [FFmpeg-devel] [PATCH] libavcodec/libopenjpegenc.c

[Paul B Mahol]

On 3/10/17, Michael Bradshaw  wrote:
> On Thu, Mar 9, 2017 at 8:09 PM, Aaron Boxer  wrote:
>>
>> Thanks, Michael. Here are those changes.
>
>
> Applied. Thanks for the patch!

AFAIK some of those could make use of BOOL.
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Re: [FFmpeg-devel] [PATCH] libavfilter/vf_detelecine: Added documentation to clarify the workings of the filter.

[Paul B Mahol]

On 3/9/17, gabriel.dalimo...@gmail.com  wrote:
> From: Gabriel D'Alimonte 
>
> Hi!
>
> I hope this patch adding code documentation to the detelecine filter will
> help with maintenance.
>
> ---
>  libavfilter/vf_detelecine.c | 115
> +---
>  1 file changed, 86 insertions(+), 29 deletions(-)
>

Idea is fine, but I do not like style of comments.
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Re: [FFmpeg-devel] [PATCH] swresample/options: enable linear_interp and exact_rational by default

[Paul B Mahol]

On 3/10/17, Michael Niedermayer  wrote:
> On Fri, Mar 10, 2017 at 11:11:05AM +0700, Muhammad Faiz wrote:
>> On Fri, Mar 10, 2017 at 10:44 AM, Michael Niedermayer
>>  wrote:
>> > On Fri, Mar 10, 2017 at 09:19:38AM +0700, Muhammad Faiz wrote:
>> >> for higher quality resampling
>> >
>> > can you hear the difference ?
>>
>> Yes, but it is sinesweep test audio, not real music audio.
>
> my question was meant to be, can you hear the difference on real
> world audio ?
>

Yes I can hear it. I made blind test.
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Re: [FFmpeg-devel] [PATCH] swresample/options: enable linear_interp and exact_rational by default

[Michael Niedermayer]

On Fri, Mar 10, 2017 at 11:11:05AM +0700, Muhammad Faiz wrote:
> On Fri, Mar 10, 2017 at 10:44 AM, Michael Niedermayer
>  wrote:
> > On Fri, Mar 10, 2017 at 09:19:38AM +0700, Muhammad Faiz wrote:
> >> for higher quality resampling
> >
> > can you hear the difference ?
> 
> Yes, but it is sinesweep test audio, not real music audio.

my question was meant to be, can you hear the difference on real
world audio ?


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Re: [FFmpeg-devel] problem with block level audio decoding on a that is not streaming

[Moritz Barsnick]

On Wed, Mar 08, 2017 at 15:32:18 -0500, Brett Tiplitz wrote:
> I've tried to update my code from avcodec_decode_audio4 to the new
> avcodec_send_packet.  I've got a block of data to decode, so this is not a
> streaming design.
> 
> 1.  When I pass a big block of data to the interface, the time required to
> process the buffer goes from 1.5s to 30s.
[...]

It's probably worth opening a ticket on trac, if you think it's a bug.
If you think it's a usage issue, you should ask on libav-user. You're
not likely to get a lot of attention here. ;-)

Moritz
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Re: [FFmpeg-devel] [PATCH] avfilter/allfilters: Prefer s32 format over float

[Moritz Barsnick]

> Because float actually int24 in the range [-1, 1]

This sentence is missing a verb and possibly an adverb or adjective. I
don't understand the point. ;-)

Moritz
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