And forward it to the underlying UDP protocol.
Fixes ticket #7517.
Signed-off-by: Jiangjie Gao <gaojiangjie@live.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Add the "http_proxy" option and its handling to the "tls" protocol,
pass the option from the "https" protocol.
The "https" protocol already defines the "http_proxy" command line
option, like the "http" protocol does. The "http" protocol properly
honors that command line option in addition to the environment
variable. The "https" protocol doesn't, because the proxy is
evaluated in the underlying "tls" protocol, which doesn't have this
option, and thus only handles the environment variable, which it
has access to.
Fixes#7223.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Marton Balint <cus@passwd.hu>
av_adler32_update() is used by av_hash_update() which will be switched
to size_t at the next bump. So it also has to be made to use size_t.
This is also necessary for framecrcenc.c, because the size of side data
will become a size_t, too.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
64 bits are needed in order to retain the uid values of Matroska
chapters; the type is kept signed because the semantics of NUT chapters
depend upon whether the id is > 0 or < 0.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This cap is currently used to mark multithreading-capable codecs that
wrap external libraries with their own multithreading code. The name is
highly confusing for our API users, since libavcodec ALWAYS handles
thread_count=0 (see commit message in previous commit). Therefore rename
the cap and update its documentation to make its meaning clear.
The old name is kept deprecated until next+1 major bump.
In the meanwhile libx264 allows to be configured for including both 8/10 bit
support within a single library. The new libx264 interface was enabled in
2f96190732.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
The option allows to select a specific window instead of the whole
screen.
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Also remove AV_LOG_SIMULATE from the list as it is not used directly, and do
not use panic level on unknown loglevel, but make them warn. Also fix mapping of
NOTICE/INFO/VERBOSE and add documentation about when the option should actually
be used.
Signed-off-by: Marton Balint <cus@passwd.hu>
Maximum packet size is 10000 (RIST_MAX_PACKET_SIZE, which is unfortunately
private) minus the RIST protocol overhead which is 28 bytes for the unencrypted
case, 36 for the encrypted case.
Signed-off-by: Marton Balint <cus@passwd.hu>
This callback is functionally the same as get_buffer2() is for decoders, and
implements for the new encode API the functionality of the old encode API had
where the user could provide their own buffers.
Reviewed-by: Lynne <dev@lynne.ee>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: James Almer <jamrial@gmail.com>
This commit adds a "gophers" handler to the gopher protocol. gophers
is a community-adopted protocol that acts the same way like normal
gopher with the added TLS encapsulation.
The gophers protocol is supported by gopher servers like geomydae(8),
and clients like curl(1), clic(1), and hurl(1).
This commit also adds compilation guards to both gopher and gophers,
since now there are two protocols in the file it makes sense to
have this addition.
Signed-off-by: parazyd <parazyd@dyne.org>
Signed-off-by: Marton Balint <cus@passwd.hu>
av_stream_add_side_data() already defines size as a size_t, so this makes it
consistent across all side data functions.
Signed-off-by: James Almer <jamrial@gmail.com>
av_packet_add_side_data() already defines size as a size_t, so this makes it
consistent across all side data functions
Signed-off-by: James Almer <jamrial@gmail.com>
Enables writing TTML documents or encoded TTML paragraphs as such
documents.
Additionally, a test for the combined TTML encoder and muxer has
been added to validate that the components still work.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
Enables the usage of such values as AV_EF_EXPLODE in encoders, which
can be useful in cases such as subtitle encoders where they have the
responsibility to validate the correctness of an incoming ASS dialog line.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
This flag was added in 492026209b
in conjunction with av_demuxer_open() to allow to pass private
options to demuxers. It worked as follows: av_open_input_stream()
(the predecessor of avformat_open_input()) would not call the
read_header function if this flag is set. Instead the user could set
private options of the demuxer via the format's private class after
avformat_open_input() and then call av_demuxer_open() which called
the format's read_header function.
This approach was abandoned in e37f161e66
and av_demuxer_open() deprecated; instead the AVDictionary based way of
passing private options to the demuxer was choosen. Yet
AVFMT_FLAG_PRIV_OPT has never been deprecated and av_demuxer_open()
never removed. This commit implements the deprecation of the flag and
schedules av_demuxer_open for removal on the next major bump.
Given that av_demuxer_open() has been deprecated in 2012 and that this
flag is useless without it, the flag will be ignored after the next
major version bump.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
AVFrame hasn't been a struct defined in libavcodec for a decade now, when
it was moved to libavutil.
Found-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The FF_API macros are private and must not be used by external callers.
As the fields in question are to be removed without replacement, just
drop them.
The fields are:
AVPacket.convergence_duration
AVCodecContext.time_base
AVCodecContext.timecode_frame_start
AV_PIX_FMT_FLAG_PSEUDOPAL pixel descriptor flag
This commit adds support for in-place FFT transforms. Since our
internal transforms were all in-place anyway, this only changes
the permutation on the input.
Unfortunately, research papers were of no help here. All focused
on dry hardware implementations, where permutes are free, or on
software implementations where binary bloat is of no concern so
storing dozen times the transforms for each permutation and version
is not considered bad practice.
Still, for a pure C implementation, it's only around 28% slower
than the multi-megabyte FFTW3 in unaligned mode.
Unlike a closed permutation like with PFA, split-radix FFT bit-reversals
contain multiple NOPs, multiple simple swaps, and a few chained swaps,
so regular single-loop single-state permute loops were not possible.
Instead, we filter out parts of the input indices which are redundant.
This allows for a single branch, and with some clever AVX512 asm,
could possibly be SIMD'd without refactoring.
The inplace_idx array is guaranteed to never be larger than the
revtab array, and in practice only requires around log2(len) entries.
The power-of-two MDCTs can be done in-place as well. And it's
possible to eliminate a copy in the compound MDCTs too, however
it'll be slower than doing them out of place, and we'd need to dirty
the input array.
This is Visual Information Fidelity (VIF) filter and one of the component
filters of VMAF. It outputs the average VIF score over all frames.
Signed-off-by: Ashish Singh <ashk43712@gmail.com>
It has been added in 6db42a2b6b,
yet since then none of the necessary create/free_device_capabilities
functions has been implemented, making this API completely useless.
Because of this one can already simplify
avdevice_capabilities_free/create and can already remove the function
pointers at the next major bump; given that the documentation explicitly
states that av_device_capabilities is not to be used by a user, it's
options can already be removed (save for the sentinel).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
A new key & value API lets us gain access to newly added parameters
without adding explicit support for them in our wrapper. Add an
option utilizing this functionality in a similar manner to other
encoder libraries' wrappers.
Signed-off-by: Bohan Li <bohanli@google.com>
In order to fine-control referencing schemes in VP9 encoding, there
is a need to use VP9E_SET_SVC_REF_FRAME_CONFIG method. This commit
provides a way to use the API through frame metadata.
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.
The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.
The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.
Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.
Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.
This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.
MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.
So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.
Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.
This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.
This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.
SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE - 10709590
DROP DSP - 10702872 - diff: -6.56KiB
DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB
DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB
SOFTCODED TABLES:
BASE - 9685096
DROP DSP - 9678378 - diff: -6.56KiB
DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB
DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB
ARM64:
HARDCODED TABLES:
BASE - 14641112
DROP DSP - 14633806 - diff: -7.13KiB
DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB
DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB
SOFTCODED TABLES:
BASE - 13636238
DROP DSP - 13628932 - diff: -7.13KiB
DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB
DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB
x86:
HARDCODED TABLES:
BASE - 12367336
DROP DSP - 12354698 - diff: -12.34KiB
DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB
DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB
SOFTCODED TABLES:
BASE - 11358094
DROP DSP - 11345456 - diff: -12.34KiB
DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB
DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB
PERFORMANCE (10min random s32le):
ARM32 - before - 39.9x - 0m15.046s
ARM32 - after - 28.2x - 0m21.525s
Speed: -30%
ARM64 - before - 36.1x - 0m16.637s
ARM64 - after - 36.0x - 0m16.727s
Speed: -0.5%
x86 - before - 184x - 0m3.277s
x86 - after - 190x - 0m3.187s
Speed: +3%
Add 2 new options:
- reconnect_on_http_error - a list of http status codes that should be
retried. the list can contain explicit status codes / the strings
4xx/5xx.
- reconnect_on_network_error - reconnects on arbitrary errors during
connect, e.g. ECONNRESET/ETIMEDOUT
the retry employs the same exponential backoff logic as the existing
reconnect/reconnect_at_eof flags.
related tickets:
https://trac.ffmpeg.org/ticket/6066https://trac.ffmpeg.org/ticket/7768
Signed-off-by: Marton Balint <cus@passwd.hu>
Do it only when requested with the AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS
flag.
Drop previous code using the long-deprecated AV_FRAME_DATA_QP_TABLE*
API. Temporarily disable fate-filter-pp, fate-filter-pp7,
fate-filter-spp. They will be reenabled once these filters are converted
in following commits.
because hls_enc_key and hls_enc_iv get 16byte char
for example:
-hls_enc_key 0123456789abcdef -hls_enc_iv abcdefghijklmnop
Reviewed-by: Gyan Doshi <ffmpeg@gyani.pro>
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
At present, progress stats are updated at a hardcoded interval of
half a second. For long processes, this can lead to bloated
logs and progress reports.
Users can now set a custom period using option -stats_period
Default is kept at 0.5 seconds.
The cropdetect filter, at present, skips the first two frames. This
behaviour is hardcoded.
New option 'skip' allows users to change this. Convenient for when
input is a single image or a trimmed video stream.
Default is kept at 2 to preserve current behaviour.
the init.mp4 can be expanded with strftime the same way as
hls_segment_filename.
Signed-off-by: Nikola Pajkovsky <nikola@pajkovsky.cz>
Signed-off-by: liuqi05 <liuqi05@kuaishou.com>
Also changes some default values for options after this change.
This makes distinction between feedback and wet option.
Before they would produce same output if values were swapped.
They add considerable complexity to frame-threading implementation,
which includes an unavoidably leaking error path, while the advantages
of this option to the users are highly dubious.
It should be always possible and desirable for the callers to make their
get_buffer2() implementation thread-safe, so deprecate this option.
This patch introduces a new frame side data type AVFilmGrainParams for use
with video codecs which support it.
It can save a lot of memory used for duplicate processed reference frames and
reduce copies when applying film grain during presentation.
This patch adds a select_region option to the xcbgrab input device.
If set to 1, the user will be prompted to select the grabbing area
graphically by clicking and dragging. A rectangle will be drawn to
mark the grabbing area. A single click with no dragging will select
the whole screen. The option overwrites the video_size, grab_x, and
grab_y options if set by the user.
For testing, just set the select_region option as follows:
ffmpeg -f x11grab -select_region 1 -i :0.0 output.mp4
The drawing happens directly on the root window using standard rubber
banding techniques, so it is very efficient and doesn't depend on any
X extensions or compositors.
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Omar Emara <mail@OmarEmara.dev>
This way the old max queue size limit based behavior for streams
where each individual packet is large is kept, while for smaller
streams more packets can be buffered (current default is at 50
megabytes per stream).
For some explanation, by default ffmpeg copies packets from before
the appointed seek point/start time and puts them into the local
muxing queue. Before, it getting utilized was much less likely
since as soon as the filter chain was initialized, the encoder
(and thus output stream) was also initialized.
Now, since we will be pushing the encoder initialization to when the
first AVFrame is decoded and filtered - which only happens after
the exact seek point is hit as packets are ignored until then -
this queue will be seeing much more usage.
In more layman's terms, this attempts to fix cases such as where:
- seek point ends up being 5 seconds before requested time.
- audio is set to copy, and thus immediately begins filling the
muxing queue.
- video is being encoded, and thus all received packets are skipped
until the requested time is hit.
then we can set the rtp read timeout instead of infinite timeout.
How to test(5s timeout):
./ffprobe -i rtp://192.168.1.67:1234?timeout=5000000
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
The manual states "there is virtually no reason to use that encoder.".
It supports less sample formats than the native encoder, is less efficient
than the native encoder and is also slower and pretty much remains untested.
libwavpack also isn't being fuzzed, which given that we plug the parameters
without any sanitizing them looks concerning.
A common pattern e.g. in libavcodec is replacing/updating buffer
references: unref old one, ref new one. This function allows simplifying
such code and avoiding unnecessary refs+unrefs if the references are
already equivalent.
Allow to set the EOF timestamp.
Also: doc/filters/testsrc*: specify the rounding of the duration option.
The changes in the ref files are right.
For filter-fps-down, the graph is testsrc2=r=7:d=3.5,fps=3.
3.5=24.5/7, so the EOF of testsrc2 will have PTS 25/7.
25/7=(10+5/7)/3, so the EOF PTS for fps should be 11/7,
and the output should contain a frame at PTS 10.
For filter-fps-up, the graph is testsrc2=r=3:d=2,fps=7,
for filter-fps-up-round-down and filter-fps-up-round-up
it is the same with explicit rounding options.
But there is no rounding: testsrc2 produces exactly 6 frames
and 2 seconds, fps converts it into exactly 14 frames.
The tests should probably be adjusted to restore them to
a useful coverage.
Expressions for option fontsize of video filter drawtext have been
supported since commit 6442e4ab3c.
Signed-off-by: Andrei Rybak <rybak.a.v@gmail.com>
Revised-by: Gyan Doshi <ffmpeg@gyani.pro>
This patch allows setting a compression ratio and to
set multiple layers. The user has to input a compression
ratio for each layer.
The per layer compression ration can be set as follows:
-layer_rates "r1,r2,...rn"
for to create 'n' layers.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Requires some extraneous top side and bottom front channels to be
defined.
According to STD-B59v2, the defined channel layout is:
- FL
- FR
- FC
- LFE1
- BL
- BR
- FLc
- FRc
- BC
- LFE2
- SiL
- SiR
- TpFL
- TpFR
- TpFC
- TpC
- TpBL
- TpBR
- TpSiL
- TpSiR
- TpBC
- BtFC
- BtFL
- BtFR
Dimensions are normally specified as width x height, and this will match
the same option to libaom-av1.
Remove the indirection through the private context at the same time.
The tile_rows/cols options currently do a confusingly different thing to
the options of the same name on other encoders like libvpx and libaom.
There is no backward-compatibility reason to implement the log2 behaviour
as there was for libaom, so just get rid of them entirely.
Threaded input can increase smoothness of e.g. x11grab significantly. Before
this patch, in order to activate threaded input the user had to specify a
"dummy" additional input, with this change it is no longer required.
Signed-off-by: Marton Balint <cus@passwd.hu>
This utility helps avoid undefined behavior when doing things like
checking how much memory we need to allocate for an image before we have
allocated a buffer.
Signed-off-by: Brian Kim <bkkim@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
remove the timeout option docs part for HTTP protocol and add
auth_type option part.
Reviewed-by: Gyan Doshi <ffmpeg@gyani.pro>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Some legacy applications such as AVI2MVE expect raw RGB bitmaps
to be stored bottom-up, whereas our RIFF BITMAPINFOHEADER assumes
they are always stored top-down and thus write a negative value
for height. This can prevent reading of these files.
Option flipped_raw_rgb added to AVI and Matroska muxers
which will write positive value for height when enabled.
Note that the user has to flip the bitmaps beforehand using other
means such as the vflip filter.
broken since:
aa5c6f382b avcodec/libaomenc: Add command-line options to control the use of partition tools
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: James Zern <jzern@google.com>
This is the only use of 'FontName' with that capitalization, as both
source-code and tests use 'Fontname'. Having consistent capitalization
makes it easier to find the relevant source from the docs.
See these examples for other uses:
libavcodec/ass_split.c:68
tests/ref/fate/sub-cc:9
We can try with the srcnn model from sr filter.
1) get srcnn.pb model file, see filter sr
2) convert srcnn.pb into openvino model with command:
python mo_tf.py --input_model srcnn.pb --data_type=FP32 --input_shape [1,960,1440,1] --keep_shape_ops
See the script at https://github.com/openvinotoolkit/openvino/tree/master/model-optimizer
We'll see srcnn.xml and srcnn.bin at current path, copy them to the
directory where ffmpeg is.
I have also uploaded the model files at https://github.com/guoyejun/dnn_processing/tree/master/models
3) run with openvino backend:
ffmpeg -i input.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=openvino:model=srcnn.xml:input=x:output=srcnn/Maximum -y srcnn.ov.jpg
(The input.jpg resolution is 720*480)
Also copy the logs on my skylake machine (4 cpus) locally with openvino backend
and tensorflow backend. just for your information.
$ time ./ffmpeg -i 480p.mp4 -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.tf.mp4
…
frame= 343 fps=2.1 q=31.0 Lsize= 2172kB time=00:00:11.76 bitrate=1511.9kbits/s speed=0.0706x
video:1973kB audio:187kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.517637%
[aac @ 0x2f5db80] Qavg: 454.353
real 2m46.781s
user 9m48.590s
sys 0m55.290s
$ time ./ffmpeg -i 480p.mp4 -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=openvino:model=srcnn.xml:input=x:output=srcnn/Maximum -y srcnn.ov.mp4
…
frame= 343 fps=4.0 q=31.0 Lsize= 2172kB time=00:00:11.76 bitrate=1511.9kbits/s speed=0.137x
video:1973kB audio:187kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.517640%
[aac @ 0x31a9040] Qavg: 454.353
real 1m25.882s
user 5m27.004s
sys 0m0.640s
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
This patch adds the control for enabling rectangular partitions, 1:4/4:1
partitions and AB shape partitions.
Signed-off-by: Wang Cao <wangcao@google.com>
Signed-off-by: James Zern <jzern@google.com>
Currently, the zoompan filter exposes a 'time' variable (missing from docs) for use in
the 'zoom', 'x', and 'y' expressions. This variable is perhaps better named
'out_time' as it represents the timestamp in seconds of each output frame
produced by zoompan. This patch adds aliases 'out_time' and 'ot' for 'time'.
This patch also adds an 'in_time' (alias 'it') variable that provides access
to the timestamp in seconds of each input frame to the zoompan filter.
This helps to design zoompan filters that depend on the input video timestamps.
For example, it makes it easy to zoom in instantly for only some portion of a video.
Both the 'out_time' and 'in_time' variables have been added in the documentation
for zoompan.
Example usage of 'in_time' in the zoompan filter to zoom in 2x for the
first second of the input video and 1x for the rest:
zoompan=z='if(between(in_time,0,1),2,1):d=1'
V2: Fix zoompan filter documentation stating that the time variable
would be NAN if the input timestamp is unknown.
V3: Add 'it' alias for 'in_time. Add 'out_time' and 'ot' aliases for 'time'.
Minor corrections to zoompan docs.
Signed-off-by: exwm <thighsman@protonmail.com>
users are getting mislead by the integer, although profile
can support both const string and integer.
http://ffmpeg.org/pipermail/ffmpeg-user/2020-June/049025.html
Also fix the order of high and main, it's not my intention.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Currently, ffmpeg inserts scale filter by default in the filter graph
to force the whole decoded stream to scale into the same size with the
first frame. It's not quite make sense in resolution changing cases if
user wants the rawvideo without any scale.
Using autoscale/noautoscale as an output option to indicate whether auto
inserting the scale filter in the filter graph:
-noautoscale or -autoscale 0:
disable the default auto scale filter inserting.
ffmpeg -y -i input.mp4 out1.yuv -noautoscale out2.yuv -autoscale 0 out3.yuv
Update docs.
Suggested-by: Mark Thompson <sw@jkqxz.net>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: U. Artie Eoff <ullysses.a.eoff@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Max region ID is 87. Also the region affects not only the G0 charset but G2 and
the national subset as well.
Signed-off-by: Marton Balint <cus@passwd.hu>
Use opaque iteration state instead of the previous child class. This
mirrors similar changes done in lavf/lavc.
Deprecate the av_opt_child_class_next() API.
The "-deinterlace" was deprecated since d7edd35, over eight years
ago.
Refer to deinterlacing filters instead.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
This allows for users who derive devices to set options for the
new device context they derive.
The main use case of this is to allow users to enable extensions
(such as surface drawing extensions) in Vulkan while deriving from
the device their frames are on. That way, users don't need to write
any initialization code themselves, since the Vulkan spec invalidates
mixing instances, physical devices and active devices.
Apart from Vulkan, other hwcontexts ignore the opts argument since they
don't support options at all (or in VAAPI and OpenCL's case, options are
currently only used for device selection, which device_derive overrides).
This change makes it possible for child encoders to define custom profile
option names which can be used for setting the AVCodecContext->profile.
Also rename unit name to something rather unique, so it won't be used elsewhere.
Signed-off-by: Marton Balint <cus@passwd.hu>
This will be used for AVCodecContext->profile. By specifying constants in the
encoders we won't have to use the common AVCodecContext options table and
different encoders can use the same profile name even with different values.
Signed-off-by: Marton Balint <cus@passwd.hu>
Both are codec properties and not encoder capabilities. The relevant
AVCodecDescriptor.props flags exist for this purpose.
Signed-off-by: James Almer <jamrial@gmail.com>
This contains encoder wrappers for H264, HEVC, AAC, AC3 and MP3.
This is based on top of an original patch by wm4
<nfxjfg@googlemail.com>. The original patch supported both encoding
and decoding, but this patch only includes encoding.
The patch contains further changes by Paweł Wegner
<pawel.wegner95@gmail.com> (primarily for splitting out the encoding
parts of the original patch) and further cleanup, build compatibility
fixes and tweaks for use with Qualcomm encoders by Martin Storsjö.
Signed-off-by: Martin Storsjö <martin@martin.st>
The Matroska muxer has always mapped the title tag to the FileDescription
element for attachments streams since support for writing attachments
was added in commit c7a63a521b. This
commit merely documents this fact.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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>
