* commit 'e46a6fb7732a7caef97a916a4f765ec0f779d195':
avconv: Check that muxing_queue exists before reading from it
Mostly noop. This was fixed in FFmpeg in 7f7c494a3.
The merge makes the cosmetics match but does not include the weird
av_log().
Merged-by: Clément Bœsch <cboesch@gopro.com>
If a subtitle packet came before the first video frame could be fully
decoded, the subtitle packet would get discarded. This puts the subtitle
into a queue instead, and processes it once the attached filter graph is
initialized.
Be more careful when an input stream encounters EOF when its filtergraph
has not been configured yet. The current code would immediately mark the
corresponding output streams as finished, while there may still be
buffered frames waiting for frames to appear on other filtergraph
inputs.
This should fix the random FATE failures for complex filtergraph tests
after a3a0230a98
This merges Libav commit 94ebf55. It was previously skipped.
This is the last filter init related Libav commit that was skipped, so
this also removes the commits from doc/libav-merge.txt.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
This is a more appropriate place for it, and will also be useful in the
following commit.
This merges Libav commit d2e56cf. It was previously skipped.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
This makes sure the actual stream parameters are used, which is
important mainly for hardware decoding+filtering cases, which would
previously require various weird workarounds to handle the fact that a
fake software graph has to be constructed, but never used.
This should also improve behaviour in rare cases where
avformat_find_stream_info() does not provide accurate information.
This merges Libav commit a3a0230. It was previously skipped.
The code in flush_encoders() which sets up a "fake" format wasn't in
Libav. I'm not sure if it's a good idea, but it tends to give
behavior closer to the old one in certain corner cases.
The vp8-size-change gives different result, because now the size of
the first frame is used. libavformat reported the size of the largest
frame for some reason.
The exr tests now use the sample aspect ratio of the first frame. For
some reason libavformat determines 0/1 as aspect ratio, while the
decoder returns the correct one.
The ffm and mxf tests change the field_order values. I'm assuming
another libavformat/decoding mismatch.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
This will be useful in the following commit, after which the muxer
timebase is not always available when encoding.
This merges Libav commit 3e265ca. It was previously skipped.
There are some changes with how/when the mux_timebase field is set,
because the Libav approach often causes a too imprecise time base
to be set. This is hard, because the muxer's write_header function
can readjust the timebase, at which point we might already have
encoded packets buffered. (It might be better to buffer them after
the encoder, instead of after all the timestamp handling logic
before muxing.)
The two FATE tests change because the output time base is raised
for subtitles. (Needed to avoid certain rounding issues in other
cases.)
Includes a minor merge fix by Mark Thompson, and
avconv: Move rescale to stream timebase before monotonisation
also by Mark Thompson <sw@jkqxz.net>.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
Some callers (like do_subtitle_out()) call this with an AVPacket that is
not refcounted. This can cause undefined behavior.
Calling av_packet_move_ref() does not make a packet refcounted if it
isn't yet. (And it can't be made to, because it always succeeds,
and can't return ENOMEM.)
Call av_packet_ref() instead to make sure it's refcounted.
I couldn't find a case that is fixed by this with the current code. But
it will fix the fate-pva-demux test with the later patches applied.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
* commit 'b55566db4c51d920a6496455bb30a608e5a50a41':
avconv: use avcodec_parameters_copy() with streamcopy
The fate-aac-autobsf-adtstoasc changes from writing an audio bitdepth
based on the sample format, which is now available.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit 'ba7397baef796ca3991fe1c921bc91054407c48b':
avconv: factor out initializing stream parameters for encoding
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Thanks to Mathieu Malaterre <malat@debian.org> for reporting the
Que/Queue typo. (https://bugs.debian.org/839542)
Reviewed-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
* commit '398f015f077c6a2406deffd9e37ff34b9c7bb3bc':
avconv: buffer the packets written while the muxer is not initialized
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '1c169782cae6c5c430ff62e7d7272dc9d0e8d527':
avconv: explicitly postpone writing the header until all streams are initialized
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Decoders have previously not used AVFrame.pts, and with the upcoming
deprecation of pkt_pts (in favor of pts), this would lead to an errorneous
interpration of timestamps.
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.
This commit is based on commit 35c8580 from Anton Khirnov <anton@khirnov.net>
which was skipped in b8945c4.
The avcodec_copy_context() call in the encode path is left in place for now
as AVStream.codec is apparently still required even after porting ffmpeg to
the new bsf API.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
