Commit 1249698e1b made
ff_mjpeg_decode_dht() call build_vlc() with a wrong (too hight)
number of codes. The reason it worked is that the lengths of the extraneous
entries is initialized to zero and ff_init_vlc_sparse() ignores codes
with a length of zero. But using a too high number of codes was
nevertheless bad, because a) the assert in build_vlc() could have been
triggered (namely if the real amount of codes is 256) and b) the loop in
build_vlc() uses initialized data (leading to Valgrind errors [1]).
Furthermore, the old code spend CPU cycles in said loop although the
result won't be used anyway.
[1]: http://fate.ffmpeg.org/report.cgi?slot=x86_64-archlinux-gcc-valgrind&time=20201008025137
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Added VDPAU to list of supported formats for VP9 420 10 and 12 bit
formats. Add VP9 10/12 Bit support for VDPAU
Signed-off-by: Philip Langdale <philipl@overt.org>
This is currently safe here, because the effective lifetime of
adaptionset_lang is parse_manifest_adaptationset() (i.e. the pointer
gets overwritten each time on entry to the function and gets freed
before exiting the function), but it is nevertheless safer to reset the
pointer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Use xmlFree instead of av_freep
snip from libxml2:
* xmlGetProp:
...
* Returns the attribute value or NULL if not found.
* It's up to the caller to free the memory with xmlFree().
According to libxml2, you are supposed to use xmlFree instead of free
on the pointer returned by it, and also using av_freep on Windows will
call _aligned_free instead of normal free, causing _aligned_free to raise
SIGTRAP and crashing ffmpeg and ffplay.
Signed-off-by: Christopher Degawa <ccom@randomderp.com>
Even if such files are invalid, they can be decoded just fine.
Also stored tiles may have bigger dimensions than displayed ones,
so do not abort decoding in such cases.
These conversion appears to be exhibiting the same rounding error as the rgbf32 formats where.
I seperated the rounding value from the 16 and 128 offsets, I think it makes it a little more clear.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
av1dec should no longer attempt to output empty frames if another decoder
was used for probing and it sucessfully set a pix_fmt ever since 05872c67a4,
so we can re-add the AV_CODEC_CAP_AVOID_PROBING cap.
Signed-off-by: James Almer <jamrial@gmail.com>
The buffers used when fragmented output is enabled have up until now not
been freed in the deinit function; they leak e.g. if one errors out of
mov_write_trailer() before one reaches the point where they are normally
written out and freed. This can e.g. happen if allocating new vos_data
fails at the beginning of mov_write_trailer().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Otherwise the old data leaks whenever extradata needs to be rewritten
(e.g. when encoding FLAC with our encoder that sends an updated
extradata packet at the end).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When remuxing an rtp hint stream (or any stream with the tag "rtp "),
the mov muxer treats this as one of the rtp hint tracks it creates
internally when ordered to do so; yet this track lacks the
AVFormatContext for the hinting rtp muxer, leading to segfaults in
mov_write_udta_sdp() if a "trak" atom is written for this stream; if not,
the stream's codecpar is freed by mov_free() as if the mov muxer owned
it (it does for the internally created "rtp " tracks), but without
resetting st->codecpar, leading to double-frees lateron. This commit
therefore ignores said tag which makes rtp hint streams unremuxable.
This fixes tickets #8181 and #8186.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The earlier code was based on the assumption that AVFrame.linesize can
not be negative.
Fixes ticket #8280.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: member access within null pointer of type 'TileGroupInfo' (aka 'struct TileGroupInfo')
Fixes: 25725/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AV1_fuzzer-5166692706287616
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: James Almer <jamrial@gmail.com>
The manual states "there is virtually no reason to use that encoder.".
It supports less sample formats than the native encoder, is less efficient
than the native encoder and is also slower and pretty much remains untested.
libwavpack also isn't being fuzzed, which given that we plug the parameters
without any sanitizing them looks concerning.
Fixes: signed integer overflow: 20 * 5184056935931942919 cannot be represented in type 'long'
Fixes: 25466/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4798660247552000
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
changes since v1:
- made into fate test
- fixed c90 warnings
- tests more intermediate formats
- tested on BE mips too
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1. Remove the assumption that the message method is TEARDOWN.
2. Don't ignore the error code of ff_rtsp_parse_streaming_commands.
Signed-off-by: Martin Storsjö <martin@martin.st>
In listen mode with UDP transport, once the sender has sent
the TEARDOWN and closed the connection, poll will indicate that
one can read from the connection (indicating that the socket has
reached EOF and should be closed by the receiver as well). In this
case, parse_rtsp_message won't try to parse the command (because
it's no longer in state STREAMING), but previously just returned
zero.
Prior to f6161fccf8, this caused
udp_read_packet to return zero, which is treated as EOF by
read_packet. But after that commit, udp_read_packet would continue
if parse_rtsp_message didn't return an explicit error code.
To keep the original behaviour from before that commit, more
explicitly return an error in parse_rtsp_message when in the wrong
state.
Fixes: #8840
Signed-off-by: Martin Storsjö <martin@martin.st>
