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>
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>
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>
Currently it's exported as AVFrame.pkt_pts, which is also the only use
for that field. The reason it is done like this is that lavc used to
export various codec-specific "timing" information in AVFrame.pts, which
is not done anymore.
Since it is confusing to the callers to have a separate field which is
used only for decoder timestamps and nothing else, deprecate pkt_pts and
use just AVFrame.pts everywhere.
The deprecated avcodec_decode_video2() and avcodec_decode_audio4()
functions called av_packet_split_side_data() on the input packets. This
is required for packets produced by libavformat with the
AVFMT_FLAG_KEEP_SIDE_DATA flag unset (which is unfortunately the
default).
The new API didn't do this yet, although it didn't matter as no decoder
supports the new API yet. The emulation layer for the old API calls the
old API functions, which took care of the splitting. Add this code to
the new API codec entrypoints too, because we shouldn't send essentially
corrupted data to decoders.
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
From Libav commit 05f66706d1.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The bits_per_raw_sample represents the number of bits of precision per sample.
The field is added at the logical place, not at the end as the code was just
recently added
This fixes the regression about losing the audio sample precision information
The change in the fate test checksum un-does the change from the merge
Previous version reviewed by: wm4 <nfxjfg@googlemail.com>
Previous version reviewed by: Dominik 'Rathann' Mierzejewski <dominik@greysector.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Carl Eugen Hoyos
wm4
Mark Thompson
Michael Niedermayer
Paul B Mahol
Pavel Koshevoy
Vittorio Giovara
Sasi Inguva
Rodger Combs
James Almer
Marton Balint
Jon Toohill
Aman Gupta
Anton Khirnov
Andrey Turkin
Peter Ross
Clément Bœsch
Hendrik Leppkes