The user has no business modifying the underlying AVCodec.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Following the same logic as 061a0c14bb, this commit turns the old encode API
into a wrapper for the new one.
Signed-off-by: James Almer <jamrial@gmail.com>
This commit follows the same logic as 061a0c14bb, but for the encode API: The
new public encoding API will no longer be a wrapper around the old deprecated
one, and the internal API used by the encoders now consists of a single
receive_packet() callback that pulls frames as required.
amf encoders adapted by James Almer
librav1e encoder adapted by James Almer
nvidia encoders adapted by James Almer
MediaFoundation encoders adapted by James Almer
vaapi encoders adapted by Linjie Fu
v4l2_m2m encoders adapted by Andriy Gelman
Signed-off-by: James Almer <jamrial@gmail.com>
Currently the frame pool used by the default get_buffer2()
implementation is a single struct, allocated when opening the decoder.
A pointer to it is simply copied to each frame thread and we assume that
no thread attempts to modify it at an unexpected time. This is rather
fragile and potentially dangerous.
With this commit, the frame pool is made refcounted, with the reference
being propagated across threads along with other context variables. The
frame pool is now also immutable - when the stream parameters change we
drop the old reference and create a new one.
Fixes: out of array access
Fixes: 18432/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAV2_fuzzer-5675574936207360
Fixes: 18326/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAV2_fuzzer-5071752362721280
Fixes: 18384/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAV1_fuzzer-5769439500304384
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2 * -1306460384 cannot be represented in type 'int'
Fixes: 17685/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_fuzzer-5747390337777664
Fixes: 17688/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INTERPLAY_ACM_fuzzer-5739287210885120
Fixes: 17699/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INTERPLAY_ACM_fuzzer-5678394531905536
Fixes: 17738/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TAK_fuzzer-5763415733174272
Fixes: 17746/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_BINKAUDIO_RDFT_fuzzer-5703008159006720
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
4650975 -> 4493240 dezicycles
This optimizes lines 2 and later. Line 1 still uses av_memcpy_backptr()
This change originally fixed ossfuzz 10790 but this is now fixed by other
optimizations already
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: NULL pointer dereference
Fixes: 15733/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IDF_fuzzer-5658616977162240
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
after a successful init if the function fails for another reason close
the codec without requiring FF_CODEC_CAP_INIT_CLEANUP which is meant to
cover init failures themselves. fixes a memory leak in those cases.
BUG=oss-fuzz:15529
Signed-off-by: James Zern <jzern@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This avoids the need for each decoder separately having to handle this case
Fixes: shift exponent -100663046 is negative
Fixes: out of array access
Fixes: 15270/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5727829913763840
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Initialize the bsfs once when opening the codec and uninitialize them once when
closing it, instead of at every codec flush/seek.
Signed-off-by: James Almer <jamrial@gmail.com>
This commit implements a full ATRAC9 decoder, a simple low-delay codec
developed by Sony and used in most PSVita games, some PS3 games and some
PS4 games. Its similar to AAC in that it uses Huffman coded scalefactors
but instead of vector quantization it just Huffman codes the spectral
coefficients (in a way similar to how Opus splits band energy coding
into coarse and fine precision). It opts to write rather large Huffman
codes by packing several small coefficients into one Huffman coded
symbol, though I don't believe this increases efficiency at all.
Band extension implements SBC in a simple way, first it mirrors the
lower spectrum onto the higher frequencies and then it uses one of 5
filters to shape it. Noise substitution is implemented via 2 of them.
Unlike previous ATRAC codecs, there's no QMF, this is a standard MDCT
codec.
Based off of the reverse engineering work of Alex Barney.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Fixes: out of array access
Fixes: poc_0411
Found-by: GwanYeong Kim <gy741.kim@gmail.com>
Tested-by: GwanYeong Kim <gy741.kim@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is just a lot of complicated and confusing code that had no purpose
anymore.
Also, the functions return values were checked only sometimes. Locking
shouldn't fail anyway, so remove the return values. Barely any other
pthread lock calls check the return value (including more important code
that is more likely to fail horribly if locking fails).
It could be argued that it might be helpful in some debugging
situations, or when the user built FFmpeg without thread support against
all good advice.
But there are dummy atomics too, so the atomic check won't help with
ensuring correctness absolutely. You gain very little.
Also, for debugging, you can just raise the ASSERT_LEVEL, and then
libavutil/thread.h will redefine the locking functions to explicitly
check the return values.
It's completely absurd that libavcodec would care about libavformat
locking, but it was there because the lock manager was in libavcodec.
This is more stright forward. Changes ABI, but we don't require ABI
compatibility currently.
Use static mutexes instead of requiring a lock manager. The behavior
should be roughly the same before and after this change for API users
which did not set the lock manager at all (except that a minor memory
leak disappears).
This reverts commit 590136e78d.
Atomics are not required for this variable, because it is protected
through the lock manager, and the use of atomics here is not compatible
with the c11 emulation wrappersi.
Fixes FATE on MSVC, among other setups which use the compat wrappers.
* commit '48bb0da050329e5111b00a12dfc154b7e78fb3a3':
lavc: Drop deprecated way of setting audio delay on encode
Merged-by: James Almer <jamrial@gmail.com>
Signed integer overflow is undefined behavior.
Detected with clang and -fsanitize=signed-integer-overflow
Signed-off-by: Vitaly Buka <vitalybuka@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '972c71e9cb63e24f57ee481e413199c7d88a8813':
lavc: add support for filtering packets before decoding
Merged-by: James Almer <jamrial@gmail.com>
* commit '549d0bdca53af7a6e0c612ab4b03baecf3a5878f':
decode: be more explicit about storing the last packet properties
Also copy pkt->size in extract_packet_props(), as it's needed for
AVFrame.pkt_size
Merged-by: James Almer <jamrial@gmail.com>
* commit '3fe2a01df7f2c193805809f57b61d79607572351':
lavc: move decoding-related code from utils.c to a new file
Merged-by: James Almer <jamrial@gmail.com>
* commit '328cd2b599bc2d0d38f3c12606fa2a66eeec016e':
lavc: move encoding-related code from utils.c to a new file
Merged-by: James Almer <jamrial@gmail.com>
This tries to handle cases where separate invocations of decode_frame()
(each running in separate threads) write to respective fields in the
same AVFrame->data[]. Having per-field owners makes interaction between
readers (the referencing thread) and writers (the decoding thread)
slightly more optimal if both accesses are field-based, since they will
use the respective producer's thread objects (mutex/cond) instead of
sharing the thread objects of the first field's producer.
In practice, this fixes the following tsan-warning in fate-h264:
WARNING: ThreadSanitizer: data race (pid=21615)
Read of size 4 at 0x7d640000d9fc by thread T2 (mutexes: write M1006):
#0 ff_thread_report_progress pthread_frame.c:569 (ffmpeg:x86_64+0x100f7cf54)
[..]
Previous write of size 4 at 0x7d640000d9fc by main thread (mutexes: write M1004):
#0 update_context_from_user pthread_frame.c:335 (ffmpeg:x86_64+0x100f81abb)
libavcodec now automatically serializes decoding for hwaccels which
are not thread-safe. This means API users, which rely on the libavcodec
native software fallback mechanism, can now simply enable threading
without running into problems.
This patch deprecates anything that has to do with merging/splitting
side data. Automatic side data merging (and splitting), as well as all
API symbols involved in it, are removed completely.
Two FF_API_ defines are dedicated to deprecating API symbols related to
this: FF_API_MERGE_SD_API removes av_packet_split/merge_side_data in
libavcodec, and FF_API_LAVF_KEEPSIDE_FLAG deprecates
AVFMT_FLAG_KEEP_SIDE_DATA in libavformat.
Since it was claimed that changing the default from merging side data to
not doing it is an ABI change, there are two additional FF_API_ defines,
which stop using the side data merging/splitting by default (and remove
any code in avformat/avcodec doing this): FF_API_MERGE_SD in libavcodec,
and FF_API_LAVF_MERGE_SD in libavformat.
It is very much intended that FF_API_MERGE_SD and FF_API_LAVF_MERGE_SD
are quickly defined to 0 in the next ABI bump, while the API symbols are
retained for a longer time for the sake of compatibility.
AVFMT_FLAG_KEEP_SIDE_DATA will (very much intentionally) do nothing for
most of the time it will still be defined. Keep in mind that no code
exists that actually tries to unset this flag for any reason, nor does
such code need to exist. Code setting this flag explicitly will work as
before. Thus it's ok for AVFMT_FLAG_KEEP_SIDE_DATA to do nothing once
side data merging has been removed from libavformat.
In order to avoid that anyone in the future does this incorrectly, here
is a small guide how to update the internal code on bumps:
- next ABI bump (probably soon):
- define FF_API_LAVF_MERGE_SD to 0, and remove all code covered by it
- define FF_API_MERGE_SD to 0, and remove all code covered by it
- next API bump (typically two years in the future or so):
- define FF_API_LAVF_KEEPSIDE_FLAG to 0, and remove all code covered
by it
- define FF_API_MERGE_SD_API to 0, and remove all code covered by it
This forces anyone who actually wants packet side data to temporarily
use deprecated API to get it all. If you ask me, this is batshit fucked
up crazy, but it's how we roll. Making AVFMT_FLAG_KEEP_SIDE_DATA to be
set by default was rejected as an ABI change, so I'm going all the way
to get rid of this once and for all.
Reviewed-by: James Almer <jamrial@gmail.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'e9bfff1cc66c85b91b262c41e8aa5e8685606225':
lavc: free buffer_frame/pkt on avcodec_open2() failure
This commit is a noop, see 27adf9f9cd.
Only reordered to reduce diff.
Merged-by: Clément Bœsch <u@pkh.me>
Allows to get a more realistic total bitrate (and estimated file size)
in avi_write_header. Previously a static default value of 200k was
assumed.
Adds an internal helper function for bitrate guessing.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Apparently the demuxer outputs the wrong padding for HE-AAC (based on
the raw sample rate, or so). aacdec contains a hack to adjust the muxer
padding accordingly before it's used to trim the decoder output. This
modified the packet side data, which in combination with the old
decoding API would change the packet the user passed to the decoder.
This is clearly not allowed, and it breaks running some gapless fate
tests with "-fflags +keepside" applied (without keepside, the packet
metadata is typically newly allocated, essentially making a copy and not
modifying the user's input packet).
This should probably be fixed in the demuxer (and consequently also the
muxer), but for now only fix the immediate problem.
Regression since 946ed78f5f (2012).
There is no reason that draining couldn't return an error or two. But
some decoders don't handle this very well, and might always return an
error. This can lead to API users getting into an infinite loop and
burning CPU, because no progress is made and EOF is never returned.
In fact, ffmpeg.c contains a hack against such a case. It is made
unnecessary with this commit, and removed with the next one. (This
particular error case seems to have been fixed since the hack was
added, though.)
This might lose frames if decoding returns errors during draining.
Fixes out of array access
Fixes: 452/fuzz-1-ffmpeg_VIDEO_AV_CODEC_ID_INTERPLAY_VIDEO_fuzzer
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The assumption that avcodec_send_packet makes regarding decoders
consuming the entire packet is not true if the codec supports
truncated decoding mode and the truncated flag is turned on.
Steps to reproduce:
./ffmpeg_g -flags truncated \
-i "http://samples.ffmpeg.org/MPEG2/test-ebu-422.40000.pakets.ts" \
-c:v ffv1 -c:a copy -y /tmp/truncated.nut
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently, the new decoding API is pretty much just a wrapper around the
old deprecated one. This is problematic, since it interferes with making
full use of the flexibility added by the new API. The old API should
also be removed at some future point.
Reorganize the code so that the new send_packet/receive_frame functions
call the actual decoding directly and change the old deprecated
avcodec_decode_* functions into wrappers around the new API.
The new internal API for decoders is now changing as well. Before this
commit, it mirrors the public API, so the decoders need to implement
send_packet() and receive_frame() callbacks. This turns out to require
awkward constructs in both the decoders and the generic code. After this
commit, the decoders only implement the receive_frame() callback and
call a new internal function, ff_decode_get_packet() to obtain input
data, in the same manner to how the bitstream filters now work.
avcodec will now always make a reference to the input packet, which means
that non-refcounted input packets will be copied. Keeping the previous
behaviour, where this copy could sometimes be avoided, would make the
code significantly more complex and fragile for only dubious gains,
since packets are typically small and everyone who cares about
performance should use refcounted packets anyway.
The current code stores a pointer to the packet passed to the decoder,
which is then used during get_buffer() for timestamps and side data
passthrough. However, since this is a pointer to user data which we do
not own, storing it is potentially dangerous. It is also ill defined for
the new decoding API with split input/output.
Fix this problem by making an explicit internally owned copy of the
packet properties.
