This way if an index table segment is present multiple times, we can always use
the proper one instead of the invalid one.

Fixes seeking in the sample of ticket #5671.
Signed-off-by: Marton Balint <cus@passwd.hu>

Signed-off-by: Mina <minasamy_@hotmail.com>

frame_number will not increase if nothing is output

Fixes: Timeout
Fixes: 9057/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CDGRAPHICS_fuzzer-4844661498707968

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>

This improves speed

Fixes: Timeout
Fixes: 9010/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TSCC_fuzzer-6042614817095680

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>

We do not want to loose the side data in case of errors
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

And add it to the CONFIGURABLE_COMPONENTS list in Makefile. This way, changes
to the new file will be tracked and the usual warning to suggest re-running
configure will be shown.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>

Fixes the following warnings:
In file included from libavcodec/fft_fixed.c:21:0:
libavcodec/fft_template.c:528:6: warning: ‘pass_big’ defined but not used [-Wunused-function]
 PASS(pass_big)
      ^
libavcodec/fft_template.c:505:13: note: in definition of macro ‘PASS’
 static void name(FFTComplex *z, const FFTSample *wre, unsigned int n)\
             ^~~~
CC      libavcodec/ffv1.o
In file included from libavcodec/fft_float.c:21:0:
libavcodec/fft_template.c:528:6: warning: ‘pass_big’ defined but not used [-Wunused-function]
 PASS(pass_big)
      ^
libavcodec/fft_template.c:505:13: note: in definition of macro ‘PASS’
 static void name(FFTComplex *z, const FFTSample *wre, unsigned int n)\
             ^~~~

GCC requires the argument to vec_splat_u32 to be a literal.  The easiest
way to accomplish this is to change 'shift' to be const in scale (as it
is in the transform routine above), and convert both routines to be
inline.  This way, GCC can coerce the values to literals.

Tested on a 970 (Apple G5) and POWER9 (Talos II); passed fate and played
a clip of Big Buck Bunny correctly.

Fixes ticket #7048
Signed-off-by: A. Wilcox <AWilcox@Wilcox-Tech.com>

A packet may have Metadata OBUs but no Sequence Header OBU, which is
useless as extradata.
Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: James Almer <jamrial@gmail.com>

aom_codec_get_global_headers() is not implemented as of libaom 1.0.0 for AV1, so
we're forced to extract the relevant header OBUs from the first packet and propagate
them as packet side data.
Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Fixes two warnings:
libavfilter/af_afir.c:194:45: warning: assuming signed overflow does not occur when assuming that (X - c) > X is always false [-Wstrict-overflow]
     int dx = FFABS(x1-x0), sx = x0 < x1 ? 1 : -1;
                                 ~~~~~~~~~~~~^~~~
libavfilter/af_aiir.c:689:45: warning: assuming signed overflow does not occur when assuming that (X - c) > X is always false [-Wstrict-overflow]
     int dx = FFABS(x1-x0), sx = x0 < x1 ? 1 : -1;
                                 ~~~~~~~~~~~~^~~~

On macOS, a zero rc_max_rate cause an error from
VTSessionSetProperty(kVTCompressionPropertyKey_DataRateLimits).

on iOS (depending on device/version), a zero rc_max_rate cause invalid
arguments from the vtenc_output_callback after few frames and then a crash
within the VideoToolbox library.
Signed-off-by: Aman Gupta <aman@tmm1.net>

Fixes a bug that would prevent using multiple comma-separated filters,
and allows options to be passed to each filter.

Based on similar loop in ffmpeg_opt.c's new_output_stream().
Signed-off-by: Aman Gupta <aman@tmm1.net>

May fix: oss fuzz issue 9220
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

This fixes out of array accesses
No testcase known
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

reget buffer can be time consuming, theres no need to do it if failures
occur

Testcase: 9010/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TSCC_fuzzer-6042614817095680

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

6 FATE references updated.

Fixes #7104

First output is audio and is rematrixed to stereo

Fixes: out of array read
Fixes: 9350/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DVDSUB_fuzzer-5746777750765568

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

No testcase
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

avcodec/lagarith: Check that the range coded data stream is consistent when the probabilities indicate no data could have been coded.

Fixes: Timeout
Fixes: 8638/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LAGARITH_fuzzer-5132046098759680
Fixes: 8943/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LAGARITH_fuzzer-4883030219948032

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>

Fixes: 8695/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-4906172426485760

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>

Lensfun is a library that applies lens correction to an image using a
database of cameras/lenses (you provide the camera and lens models, and
it uses the corresponding database entry's parameters to apply lens
correction). It is licensed under LGPL3.

The lensfun filter utilizes the lensfun library to apply lens
correction to videos as well as images.

This filter was created out of necessity since I wanted to apply lens
correction to a video and the lenscorrection filter did not work for me.

While this filter requires little info from the user to apply lens
correction, the flaw is that lensfun is intended to be used on indvidual
images. When used on a video, the parameters such as focal length is
constant, so lens correction may fail on videos where the camera's focal
length changes (zooming in or out via zoom lens). To use this filter
correctly on videos where such parameters change, timeline editing may
be used since this filter supports it.

Note that valgrind shows a small memory leak which is not from this
filter but from the lensfun library (memory is allocated when loading
the lensfun database but it somehow isn't deallocated even during
cleanup; it is briefly created in the init function of the filter, and
destroyed before the init function returns). This may have been fixed by
the latest commit in the lensfun repository; the current latest release
of lensfun is almost 3 years ago.

Bi-Linear interpolation is used by default as lanczos interpolation
shows more artifacts in the corrected image in my tests.

The lanczos interpolation is derived from lenstool's implementation of
lanczos interpolation. Lenstool is an app within the lensfun repository
which is licensed under GPL3.

v2 of this patch fixes license notice in libavfilter/vf_lensfun.c

v3 of this patch fixes code style and dependency to gplv3 (thanks to
Paul B Mahol for pointing out the mentioned issues).

v4 of this patch fixes more code style issues that were missed in
v3.

v5 of this patch adds line breaks to some of the documentation in
doc/filters.texi (thanks to Gyan Doshi for pointing out the issue).

v6 of this patch fixes more problems (thanks to Moritz Barsnick for
pointing them out).

v7 of this patch fixes use of sqrt() (changed to sqrtf(); thanks to
Moritz Barsnick for pointing this out). Also should be rebased off of
latest master branch commits at this point.
Signed-off-by: Stephen Seo <seo.disparate@gmail.com>

MP3 frames may not be aligned to aa chunk boundaries. When seeking,
calculate the expected frame offset in the target chunk. Adjust the
timestamp and truncate the next packet accordingly.

This solution works for the majority of tested audio material. For
some rare encodings with mp3 padding or embedded id3 tags, it will
mispredict the correct offset, and at worst skip an extra frame.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>