Full width text is really difficult to read, this makes it more
more legible on larger (widescreen) screens. It also means we aren't
inventing our own container instead of using the bootstrap one.
Signed-off-by: Josh de Kock <josh@itanimul.li>
There is really no need for two aac wrappers, we already have
libfdk-aac which is better. Not to mention that faac doesn't
even support HEv1, or HEv2. It's also under a license which is
unusable for distribution, so it would only be useful to people
who will compile their own ffmpeg, only use it themselves (which
at that point should just use fdk-aac).
Signed-off-by: Josh de Kock <josh@itanimul.li>
This is a bit messy, mainly due to timestamp handling.
decode_video() relied on the fact that it could set dts on a flush/drain
packet. This is not possible with the old API, and won't be. (I think
doing this was very questionable with the old API. Flush packets should
not contain any information; they just cause a FIFO to be emptied.) This
is replaced with checking the best_effort_timestamp for AV_NOPTS_VALUE,
and using the suggested DTS in the drain case.
The modified tests (fate-cavs and others) still fails due to dropping
the last frame. This happens because the timestamp of the last frame
goes backwards (ffprobe -show_frames shows the same thing). I suspect
that this "worked" due to the best effort timestamp logic picking the
DTS over the decreasing PTS. Since this logic is in libavcodec (where
it probably shouldn't be), this can't be easily fixed. The timestamps
of the cavs samples are weird anyway, so I chose not to fix it.
Another strange thing is the timestamp handling in the video path of
process_input_packet (after the decode_video() call). It looks like
the code to increase next_dts and next_pts should be run every time
a frame is decoded - but it's needed even if output is skipped.
With the new decode API, doing this in ffmpeg.c is impractical. There
was resistance against removing the warning, so put it into libavcodec.
Not bothering with reducing the warning to verbose log level for
subsequent wanrings. The warning should be rare, and only happen when
developing new codecs for the old API.
Includes a change suggested by Michael Niedermayer.
* commit 'ac7bfd69678f3966e38debdb27f4bde94dc0345c':
lavfi: add a QSV scaling filter
This is a noop since it depends on sharing a hwcontext with the
decoder/encoder, see 04b17ff and 130e1f1
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit 'ad9c9440d592e4d53d6bec9961b4b22e25387d70':
qsvenc: support getting the session from an AVHWFramesContext
This commit is a noop, as it needs to be fully re-implemented for our
qsv components.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit 'a0524d9b1e1bb0012207584f067096df7792df6c':
qsvdec: support getting the session from an AVHWFramesContext
This commit is a noop, as it needs to be fully re-implemented for our qsv
components.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
when use use_localtime_mkdir to create multi level dir,
ffmpeg give error message:
ffmpeg -re -i ~/Movies/objectC/facebook.mp4 -c copy -use_localtime 1
-use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d/%s.ts'
out.m3u8
error message:
Could not create directory 20160926/file-20160926 with use_localtime_mkdir
add mkdir_p for support the multi level dir
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Tested-by: Zuo Genyu <1515161258@qq.com> (Windows)
Signed-off-by: Steven Liu <lingjiujianke@gmail.com>
Copy interrupt callback to slave format context to allow
user to interrupt IO. Copy format flags as well.
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Streaming servers appear to ignore all other language metadata.
Signed-off-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Josh de Kock <josh@itanimul.li>
Fixes gapless decoding. Adjust skip_samples field correctly in case of DISCARDed audio frames.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The check could previously never evaluate to true, probably due to
a typo.
Reported-By: Mihai Chindea <mihai.chindea@uti.eu.com>
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Tested-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Lou Logan <lou@lrcd.com>
Found-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This way, in case of bit rate not being set, max_bitrate will be
used instead. This enables, for example, re-using max_bitrate
information from the input or doing transcoding with a rate
control mode that is not bit rate based.
Signed-off-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Timo Rothenpieler
Matthieu Bouron
James Almer
Michael Niedermayer
Josh de Kock
Marton Balint
Carl Eugen Hoyos
wm4
Hendrik Leppkes
Clément Bœsch
Steven Liu
Jan Sebechlebsky
Anton Khirnov
Jan Ekström
Sasi Inguva
Sasi Inguva
Moritz Barsnick