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>
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>
