It does not work correctly and apparently never did. There is no
indication that this (mis)feature is ever used in the wild or even that
any software other than the reference supports it.
Since the code that attempts to support it adds some nontrivial
complexity and has resulted in several bugs in the past, it is better to
just drop it.
Currently, it needs to be initialized by the ER caller (which is
currently either a mpegvideo decoder or h264dec). However, since none of
those decoders use MECmpContext for anything except ER, it makes more
sense to handle it purely inside ER.
The frame size must be set by the caller and each dimension must be a
multiple of 2.
CC: libav-stable@libav.org
Bug-ID: CVE-2014-8543
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
The frame size must be set by the caller and each dimension must be a
multiple of 8.
CC: libav-stable@libav.org
Bug-ID: CVE-2014-8542
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Fixes possible invalid memory access.
Based on code by Michael Niedermayer <michaelni@gmx.at>
CC: libav-stable@libav.org
Bug-ID: CVE-2014-8541
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Also add no-op fallbacks when threading is disabled.
This helps keeping the code clean if Libav is compiled for targets
without threading. Since we assume that no threads of any kind are used
in such configurations, doing nothing is ok by definition.
Based on a patch by wm4 <nfxjfg@googlemail.com>.
The only parameters needed by the demuxers are the sample rate and sample
count, which can be trivially extracted manually, without resorting to
an avpriv function.
It will not be set unless the codec context is used as the encoding
context, which is discouraged. For MP2, av_get_audio_frame_duration()
will already set the frame size properly. For MP3, set the frame size
explicitly.
Currently, the API takes an external AVCodecContext, which is used only
for extradata and logging. This change will allow to it to work without
an AVCodecContext in the following commits.
The quality scale field is only supposed to be present if the fourth bit
is set. In practice, lame always sets it, but other tools might not.
CC:libav-stable@libav.org
Currently, this option is accessed through AVCodecContext.mb_threshold,
which originally controlled reusing MB data when transcoding mpeg to
mpeg. Since the libvpx meaning is completely different from the original
mpegvideo meaning, it is better to use a separate private option for
this.
When decoding, this field holds the inverse of the framerate that can be
written in the headers for some codecs. Using a field called 'time_base'
for this is very misleading, as there are no timestamps associated with
it. Furthermore, this field is used for a very different purpose during
encoding.
Add a new field, called 'framerate', to replace the use of time_base for
decoding.
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
codec context is used for encoding or decoding (and has yet another
different meaning for video), preventing generic handling of the codec
context.
Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
At least one FATE sample contains such chunks and happens to work simply
by accident (due to find_stream_info() swallowing the error).
CC: libav-stable@libav.org
It is basically a wrapper around av_get_audio_frame_duration(), with a
fallback to AVCodecContext.frame_size. However, that field is set only
when the stream codec context is actually used for encoding or decoding,
which is discouraged.
For muxing, it is generally the responsibility of the caller to set the
packet duration.
For demuxing, if the duration is not stored at the container level, it
should be set by the parser.
Therefore, removing the frame_size fallback should not break any
important case.
(cherry picked from commit 30e50c5027)
Conflicts:
libavformat/utils.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is basically a wrapper around av_get_audio_frame_duration(), with a
fallback to AVCodecContext.frame_size. However, that field is set only
when the stream codec context is actually used for encoding or decoding,
which is discouraged.
For muxing, it is generally the responsibility of the caller to set the
packet duration.
For demuxing, if the duration is not stored at the container level, it
should be set by the parser.
Therefore, removing the frame_size fallback should not break any
important case.
1) each of the loops run within a single CTB, so the relevant reference
list is constant
2) when that CTB is, or lies on the same slice as, the current one, we
can use a simple access instead of a relatively expensive call to
ff_hevc_get_ref_list()
The input data must remain constant, make a copy instead. This is in
theory a performance hit, but since I failed to find any samples
using this feature, this should not matter in practice.
Also, check the size of the header, avoiding invalid reads on truncated
data.
CC:libav-stable@libav.org
Anton Khirnov
