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 27adf9f9cdbc6f1dc0a1292dc97490eec6f2f068.
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 946ed78f5f8 (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.
It's container level information on some formats (Matroska, MXF, yuv4mpeg), so
it should be printed at higher log levels than debug.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '32c8359093d1ff4f45ed19518b449b3ac3769d27':
lavc: export the timestamps when decoding in AVFrame.pts
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
With the new decode API, doing this in ffmpeg.c is impractical. There
was resistance against removing the warning, so put it into libavcodec.
Not bothering with reducing the warning to verbose log level for
subsequent wanrings. The warning should be rare, and only happen when
developing new codecs for the old API.
Includes a change suggested by Michael Niedermayer.
This makes it easier to use the lowres option when dealing with input
files in different codecs. If the codec doesn't support lowres=1 for
instance, it will throw a warning and use lowres=0 instead of erroring
out completely.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
