This option adds a long string of numbers to the progress line, where
i-th number contains the base-2 logarithm of the number of times a frame
with this QP value was seen by print_report().
There are multiple problems with this feature:
* despite this existing since 2005, web search shows no indication
that it was ever useful for any meaningful purpose;
* the format of what is printed is entirely undocumented, one has to
find it out from the source code;
* QP values above 31 are silently ignored;
* it only works with one video stream;
* as it relies on global state, it is in conflict with ongoing
architectural changes.
It then seems that the nontrivial cost of maintaining this option is not
worth its negligible (or possibly negative - since it pollutes the
already large option space) value.
Users who really need similar functionality can also implement it
themselves using -vstats.
In particular, add a sentence to introduce the example, and add a
simpler starting example with no options.
Also use different format for input.avi and output.mp4, to convey
that the conversion also works on the container format.
Address issue:
http://trac.ffmpeg.org/ticket/8730
Reference drawtext textfile option and ffmpeg -filter_complex_script
and -filter_script as possible solutions to avoid shell escaping.
Address issue:
http://trac.ffmpeg.org/ticket/9008
Analogous to -enc_stats*, but happens right before muxing. Useful
because bitstream filters and the sync queue can modify packets after
encoding and before muxing. Also has access to the muxing timebase.
Splits the currently handled subtitle at random access point
packets that can be configured to follow a specific output stream.
Currently only subtitle streams which are directly mapped into the
same output in which the heartbeat stream resides are affected.
This way the subtitle - which is known to be shown at this time
can be split and passed to muxer before its full duration is
yet known. This is also a drawback, as this essentially outputs
multiple subtitles from a single input subtitle that continues
over multiple random access points. Thus this feature should not
be utilized in cases where subtitle output latency does not matter.
Co-authored-by: Andrzej Nadachowski <andrzej.nadachowski@24i.com>
Co-authored-by: Bernard Boulay <bernard.boulay@24i.com>
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
This enables overriding the rotation as well as horizontal/vertical
flip state of a specific video stream on the input side.
Additionally, switch the singular test that was utilizing the rotation
metadata to instead override the input display rotation, thus leading
to the same result.
It has been deprecated in favor of the aresample filter for almost 10
years.
Another thing this option can do is drop audio timestamps and have them
generated by the encoding code or the muxer, but
- for encoding, this can already be done with the setpts filter
- for muxing this should almost never be done as timestamp generation by
the muxer is deprecated, but people who really want to do this can use
the setts bitstream filter
The -shortest option (which finishes the output file at the time the
shortest stream ends) is currently implemented by faking the -t option
when an output stream ends. This approach is fragile, since it depends
on the frames/packets being processed in a specific order. E.g. there
are currently some situations in which the output file length will
depend unpredictably on unrelated factors like encoder delay. More
importantly, the present work aiming at splitting various ffmpeg
components into different threads will make this approach completely
unworkable, since the frames/packets will arrive in effectively random
order.
This commit introduces a "sync queue", which is essentially a collection
of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
and are then released for further processing (encoding or muxing) when
it is ensured that the frame in question will not cause its stream to
get ahead of the other streams (the logic is similar to libavformat's
interleaving queue).
These sync queues are then used for encoding and/or muxing when the
-shortest option is specified.
A new option – -shortest_buf_duration – controls the maximum number of
queued packets, to avoid runaway memory usage.
This commit changes the results of the following tests:
- copy-shortest[12]: the last audio frame is now gone. This is
correct, since it actually outlasts the last video frame.
- shortest-sub: the video packets following the last subtitle packet are
now gone. This is also correct.
This is a per-file input option that adjusts an input's timestamps
with reference to another input, so that emitted packet timestamps
account for the difference between the start times of the two inputs.
Typical use case is to sync two or more live inputs such as from capture
devices. Both the target and reference input source timestamps should be
based on the same clock source.
If either input lacks starting timestamps, then no sync adjustment is made.
-shortest stops 'recording' when the shortest output stream ends. The
native or even seek-adjusted duration of the source input stream isn't
considered.
Currently, the code doing this is spread over several places and may
behave in unexpected ways. E.g. automatic 'default' marking is only done
for streams fed by complex filtergraphs. It is also applied in the order
in which the output streams are initialized, which is effectively
random.
Move processing the dispositions at the end of open_output_file(), when
we already have all the necessary information.
Apply the automatic default marking only if no explicit -disposition
options were supplied by the user, and apply it to the first stream of
each type (excluding attached pics) when there is more than one stream
of that type and no default markings were copied from the input streams.
Explicitly document the new behavior.
Changes the results of some tests, where the output file gets a default
disposition, while it previously did not.
UPD: Rebase of last patch set over current master and use DX9 as default device type.
Makes selection of dxva2/DX9 device type by default as before with explicit d3d11va/DX11 usage to cover more HW configurations.
Added warning message to expect changing default device type in the future.
Fixes TGL / AV1 decode as requires DX11 with explicit DX11 type
selection.
Add headless/multi adapter support and fixes:
https://trac.ffmpeg.org/ticket/7511https://trac.ffmpeg.org/ticket/6827http://ffmpeg.org/pipermail/ffmpeg-trac/2017-November/041901.htmlhttps://trac.ffmpeg.org/ticket/7933338fbcd5bbhttps://github.com/jellyfin/jellyfin/issues/2626#issuecomment-602153952
Any other fixes are welcome including OpenCL interop patch since I don't have proper setup to validate this use case
Decoding, encoding, transcoding have been validated.
child_device_type option is responsible for d3d11va/dxva2 device selection
Usage examples:
DirectX 11:
-init_hw_device qsv:hw,child_device_type=d3d11va
-init_hw_device qsv:hw,child_device_type=d3d11va,child_device=0
OR
-init_hw_device d3d11va=dx -init_hw_device qsv@dx
DirectX 9 is still supported but requires explicit selection:
-init_hw_device qsv:hw,child_device_type=dxva2
OR
-init_hw_device dxva2=dx -init_hw_device qsv@dx
Signed-off-by: Artem Galin <artem.galin@intel.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.
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.
Tobias Rapp
Davy Durham
Anton Khirnov
Sil Vilerino
Stefano Sabatini
Jan Ekström
Jan Ekström
Gyan Doshi
Marton Balint
NetSysFire
Timo Rothenpieler
Artem Galin
Keyun Tong