Here the packet size is known before allocating the packet because
the encoder provides said information (and works with internal buffers
itself), so one can use this information to avoid the implicit use of
another intermediate buffer for the packet data; and by switching to
ff_get_encode_buffer() one can also allow user-supplied buffers.
Reviewed-by: James Almer <jamrial@gmail.com>
Reviewed-by: James Zern <jzern@google.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It has been added in 2016 when this flag made no sense for encoders at
all; now that it makes sense, audiotoolboxenc doesn't support it,
despite claiming to do so.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
ref_frame is owned by the framesync structure and should therefore not
be modified; furthermore, these properties that are copied don't seem to
be used at all, so copying is unnecessary. Finally copying when the
destination frame is NULL gives a guaranteed segfault.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is marginally slower, but correct for all input values.
The previous implementation failed with certain input seeds, e.g.
"checkasm --test=hevc_idct 98".
Signed-off-by: Martin Storsjö <martin@martin.st>
The twoloop coder is highly loaded with (pseudo-)perceptual metrics,
and the aim of the tests is to piece-wise test each function of the
encoder, for which the 'fast' coder is perfect, since it only decides
on which scalefactors to use, rather than enable or disable encoder
features.
This used to be the default, but was reverted as it was slower than
the 'fast' coder by around 25%.
Since our encoder is still not very good, change back to the twoloop
coder by default. It has much better rate control management as well,
making it closer to CBR, and it sounds much better.
With some minor changes by Marton Balint:
- removed trailing whitespace
- fixed network_descriptors_length
- fixed reserved_future_use flag in the start of the section
- removed unused program variable
- emit first NIT after PAT
- some other cosmetics
Signed-off-by: Ubaldo Porcheddu <ubaldo@eja.it>
Signed-off-by: Marton Balint <cus@passwd.hu>
Also use helper function to set the timestamp. Maybe we could also use
nanosecond precision, but there were some float rounding concerns.
Signed-off-by: Marton Balint <cus@passwd.hu>
Previously, only the size of a given tile was passed, making the
offset and size marked in VASliceParameterBufferAV1 invalid with
multiple tiles.
Signed-off-by: Fei Wang <fei.w.wang@intel.com>
Fixes: CID1398579 Dereference before null check
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Trivial for an encoder that has a good estimate of the size of
the output packet in advance.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Trivial for an encoder that has a very good estimate of the size
of the output packet in advance.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Now that the proper buffer size is calculated (and checked) before
allocating the buffer, it is known that the buffer always suffices.
So use the unchecked PutBit-API; and also use an unchecked bitstream
reader as we check ourselves.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Up until now, the JPEG-LS encoder allocated a worst-case-sized packet
at the beginning of each encode2 call; then it wrote the packet header
into its destination buffer and encoded the actual packet data;
said data is written into another worst-case-sized buffer, because it
needs to be escaped before being written into the packet buffer.
Finally, because the packet buffer is worst-case-sized, the generic
code copies the actually used part into a fresh buffer.
This commit changes this: Allocating the packet and writing the header
into it is deferred until the actual data has been encoded and its size
is known. This gives a good upper bound for the needed size of the packet
buffer (the upper bound might be 1/15 too large) and so one can avoid the
implicit intermediate buffer and support user-supplied buffers by using
ff_get_encode_buffer().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This reverts commit b5ca8f2c66.
This commit will make new problem about tickets: 9193,9205
It flush data into file with init file context together,
and it can get keyframe size, maybe need more method to get keyframe
size.
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
Exists since 8a129077cc.
Fixes a -Winitializer-overrides warning when building with Clang.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Since a247ac640d, allcodecs.c contained
two lines that matched the regex used by find_filters_extern in
configure; as a result, libx264 appeared twice the list of codecs
(if enabled).
Fix this by using only one matching line by adding a preprocessor define
for the part that differed in the two old lines: The const qualifier.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Regression since b9c5fdf6027010d15ee90a43aa023e45a5189097;
fixes Coverity ID #1484786.
Also remove the check for st->internal->parser as av_parser_close(NULL)
is a no-op.
Reviewed-by: James Almer <jamrial@gmail.com>
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Otherwise the rate emulation logic in `transcode_step` never gets
hit, and the unavailability flag never gets reset, leading to an
eternal loop with some rate emulation use cases.
This change was missed during the rework of ffmpeg.c, in which
encoder initialization was moved further down the time line in
commit 67be1ce0c6 . Previously,
as the encoder initialization had happened earlier, this state was
not possible (flow getting as far as hitting the rate emulation logic,
yet not having the encoder initialized yet).
Fixes#9160
Two modes are supported in guided filter, basic mode and fast mode.
Basic mode is the initial pushed guided filter without optimization.
Fast mode is implemented based on the basic one by sub-sampling method.
The sub-sampling ratio which can be defined by users controls the
algorithm complexity. The larger the sub-sampling ratio, the lower
the algorithm complexity.
Signed-off-by: Xuewei Meng <xwmeng96@gmail.com>
Reviewed-by: Steven Liu <liuqi05@kuaishou.com>
