Two kinds of errors can happen when working with dynamic buffers:
(Re)allocation errors or truncation errors (one has to truncate the
buffer to a size of INT_MAX because avio_close_dyn_buf() and
avio_get_dyn_buf() both return an int). Right now, avio_get_dyn_buf()
returns an empty buffer in either case. But given that
avio_get_dyn_buf() does not destroy the dynamic buffer, one can return
the buffer in case of truncation and let the user check the error flags
and decide for himself instead of hardcoding a single way to proceed
in case of truncation.
(This actually restores the behaviour from before commit
163bb9ac0af495a5cb95441bdb5c02170440d28c.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This has originally been done in 568e18b15e
as a precaution against integer overflows, but it is actually easy to
support the full range of int without overflows.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If adding two ints overflows, it doesn't matter whether the result will
be stored in an unsigned or not; and checking afterwards does not make it
retroactively defined.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Use opaque iteration state instead of the previous child class. This
mirrors similar changes done in lavf/lavc.
Deprecate the av_opt_child_class_next() API.
current_picture was not writable here because a reference existed in
at least avctx->coded_frame, and potentially elsewhere if the caller
created new ones from it.
Signed-off-by: James Almer <jamrial@gmail.com>
The "-deinterlace" was deprecated since d7edd35, over eight years
ago.
Refer to deinterlacing filters instead.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
It is a constant known at codec init, so set it in
ff_frame_thread_init(). Also, only set it for video, since the meaning
of this field is not well-defined for audio with frame threading.
Fixes availability of delay in callbacks invoked from the per-thread
contexts after 1f4cf92cfb.
This causes regressions in end to end timestamps with mp3s and ffmpeg.
The revert is to avoid this regression in the 4.3 release
See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted.
This reverts commit 460132c998.
This makes got_output consistent with the code in slice_end() which sets the output
in slice_end()
if (s->pict_type == AV_PICTURE_TYPE_B || s->low_delay) {
int ret = av_frame_ref(pict, s->current_picture_ptr->f);
...
} else {
Fixes: assertion failure
Fixes: 22178/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG1VIDEO_fuzzer-5664234440753152
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This causes indexes into scale_conversion_table to wrap around, alternatively they
could be clipped, the table be enlarged or we can error out. I have not found a document that specifies
what is the correct way to handle this
Fixes: out of array access
Fixes: 21727/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HCA_fuzzer-5752477891952640.fuzz
Fixes: 22438/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HCA_fuzzer-5640717790871552
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: division by zero
Fixes: 22974/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PFM_fuzzer-6270027077779456
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
