This is intended to replace the deprecated the AV_FRAME_DATA_QP_TABLE*
API and extend it to a wider range of codecs.
In the future, it may also be extended to support other encoding
parameters such as motion vectors.
Additional changes by Anton Khirnov <anton@khirnov.net> with suggestions
by Lynne <dev@lynne.ee>.
Signed-off-by: Juan De León <juandl@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This solves a huge oversight - it lets users reliably use their own
AVVulkanDeviceContext. Otherwise, the extensions supplied and enabled
are not discoverable by anything outside of hwcontext_vulkan.
Also clarifies that any user-supplied VkInstance must be at least 1.1.
Also documents all options supported by the hwdevice.
This lets users enable all extensions they need without writing their own
instance initialization code.
After this claim was made in e34e361 kamedo2 did an in-depth ABX
test comparing these encoders:
https://hydrogenaud.io/index.php?topic=111085.0
Result: FFmpeg AAC wasn't as good as libfdk_aac on average.
I know some things have changed since then such as, "use the fast
coder as the default" (fcb681ac) for example, so maybe the situation
is different now.
However, I am unaware of any recent comparison. So without any
substantiation we shouldn't make such a blantant claim.
Signed-off-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Gyan Doshi <ffmpeg@gyani.pro>
It's based on the following specs:
RDD 45:2017 - SMPTE Registered Disclosure Doc - Interoperable Master Format - Application ProRes
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
It's based on the following specs:
RDD 36:2015 - SMPTE Registered Disclosure Doc - Apple ProRes Bitstream Syntax and Decoding Process
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Sequence numbers of segments should be unique, if an encoder is using shorter
than 1 second segments and it is restarted, then future segments will be using
already used sequence numbers if initial sequence number is based on the number
of seconds since epoch and not microseconds.
Signed-off-by: Marton Balint <cus@passwd.hu>
Because not every user know about in_pad and out_pad reasonable value range
so maybe try to set 1.0, but setting 1.0 is so hugh to get an fatal error.
Suggested-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
bump minor version for DOVI sidedata, because added the dovi_meta.h
as lavu API part. Also update APIchanges.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Up until now, the Matroska muxer would mark a track as default if it had
the disposition AV_DISPOSITION_DEFAULT or if there was no track with
AV_DISPOSITION_DEFAULT set; in the latter case even more than one track
of a kind (audio, video, subtitles) was marked as default which is not
sensible.
This commit changes the logic used to mark tracks as default. There are
now three modes for this:
a) In the "infer" mode the first track of every type (audio, video,
subtitles) with default disposition set will be marked as default; if
there is no such track (for a given type), then the first track of this
type (if existing) will be marked as default. This behaviour is inspired
by mkvmerge. It ensures that the default flags will be set in a sensible
way even if the input comes from containers that lack the concept of
default flags. This mode is the default mode.
b) The "infer_no_subs" mode is similar to the "infer" mode; the
difference is that if no subtitle track with default disposition exists,
no subtitle track will be marked as default at all.
c) The "passthrough" mode: Here the track will be marked as default if
and only the corresponding input stream had disposition default.
This fixes ticket #8173 (the passthrough mode is ideal for this) as
well as ticket #8416 (the "infer_no_subs" mode leads to the desired
output).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Previously, there was no way to flush an encoder such that after
draining, the encoder could be used again. We generally suggested
that clients teardown and replace the encoder instance in these
situations. However, for at least some hardware encoders, the cost of
this tear down/replace cycle is very high, which can get in the way of
some use-cases - for example: segmented encoding with nvenc.
To help address that use case, we added support for calling
avcodec_flush_buffers() to nvenc and things worked in practice,
although it was not clearly documented as to whether this should work
or not. There was only one previous example of an encoder implementing
the flush callback (audiotoolboxenc) and it's unclear if that was
intentional or not. However, it was clear that calling
avocdec_flush_buffers() on any other encoder would leave the encoder in
an undefined state, and that's not great.
As part of cleaning this up, this change introduces a formal capability
flag for encoders that support flushing and ensures a flush call is a
no-op for any other encoder. This allows client code to check if it is
meaningful to call flush on an encoder before actually doing it.
I have not attempted to separate the steps taken inside
avcodec_flush_buffers() because it's not doing anything that's wrong
for an encoder. But I did add a sanity check to reject attempts to
flush a frame threaded encoder because I couldn't wrap my head around
whether that code path was actually safe or not. As this combination
doesn't exist today, we'll deal with it if it ever comes up.
The current design, where
- proper init is called for the first per-thread context
- first thread's private data is copied into private data for all the
other threads
- a "fixup" function is called for all the other threads to e.g.
allocate dynamically allocated data
is very fragile and hard to follow, so it is abandoned. Instead, the
same init function is used to init each per-thread context. Where
necessary, AVCodecInternal.is_copy can be used to differentiate between
the first thread and the other ones (e.g. for decoding the extradata
just once).
currently, the model outputs the rain, and so need a subtraction
in filter c code to get the final derain result.
I've sent a PR to update the model file and accepted, see at
https://github.com/XueweiMeng/derain_filter/pull/3
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Related to this are the following changes:
* Mention the GNUmakefile that AviSynth+ provides for installing
just the headers.
* Expand on users installing AviSynth on their system a little
more.
When the user opted to write the Cues at the beginning, the Cues were
simply written without checking in advance whether enough space has been
reserved for them. If it wasn't enough, the data following the space
reserved for the Cues was simply overwritten, corrupting the file.
This commit changes this by checking whether enough space has been
reserved for the Cues before outputting anything. If it isn't enough,
no Cues will be output at all and the file will be finalized normally,
yet writing the trailer will nevertheless return an error to notify
the user that his wish of having Cues at the front of the file hasn't
been fulfilled.
This change opens new usecases for this option: It is now safe to use
this option to e.g. record live streams or to use it when muxing the
output of an expensive encoding, because when the reserved space turns
out to be insufficient, one ends up with a file that just lacks Cues
but is otherwise fine.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit updates the documentation of av_read_frame() to match its
actual behaviour in several ways:
1. On success, av_read_frame() always returns refcounted packets.
2. It can handle uninitialized packets.
3. On error, it always returns blank packets.
This will allow callers to not initialize or unref unnecessarily.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, it was completely unspecified what the content of the
destination packet dst was on error. Depending upon where the error
happened calling av_packet_unref() on dst might be dangerous.
This commit changes this by making sure that dst is blank on error, so
unreferencing it again is safe (and still pointless). This behaviour is
documented.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
They use non-public functions, which is unacceptable for a public API
example. Rename the example back to avio_list_dir.
This effectively reverts c84d208c27 and
767d780ec0.
The Y channel is handled by dnn, and also resized by dnn. The UV channels
are resized with swscale.
The command to use espcn.pb (see vf_sr) looks like:
./ffmpeg -i 480p.jpg -vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y -y tmp.espcn.jpg
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Reviewed-by: Pedro Arthur <bygrandao@gmail.com>
Only the Y channel is handled by dnn, the UV channels are copied
without changes.
The command to use srcnn.pb (see vf_sr) looks like:
./ffmpeg -i 480p.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.jpg
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Reviewed-by: Pedro Arthur <bygrandao@gmail.com>
Supports connecting to a RabbitMQ broker via AMQP version 0-9-1.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
rw_timeout is the generic URLcontext option, not the protocol specific timeout
option, also ?rw_timeout never worked because ?timeout was parsed instead.
Signed-off-by: Marton Balint <cus@passwd.hu>
This adds a decoder for Broderbund's sprite-based QuickTime CDToons
codec, based on the decoder I wrote for ScummVM.
Signed-off-by: Alyssa Milburn <amilburn@zall.org>
Required minimal changes to the code so made sense to implement.
FFT and MDCT tested, the output of both was properly rounded.
Fun fact: the non-power-of-two fixed-point FFT and MDCT are the fastest ever
non-power-of-two fixed-point FFT and MDCT written.
This can replace the power of two integer MDCTs in aac and ac3 if the
MIPS optimizations are ported across.
Unfortunately the ac3 encoder uses a 16-bit fixed point forward transform,
unlike the encoder which uses a 32bit inverse transform, so some modifications
might be required there.
The 3-point FFT is somewhat less accurate than it otherwise could be,
having minor rounding errors with bigger transforms. However, this
could be improved later, and the way its currently written is the way one
would write assembly for it.
Similar rounding errors can also be found throughout the power of two FFTs
as well, though those are more difficult to correct.
Despite this, the integer transforms are more than accurate enough.
Compared to ad-hoc if(printed) ... code this allows the user to disable
it by adjusting the log level
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
