It is wrong/incorrect in two aspects:
1. The pixel format is not enough to guarantee that the resulting file
will be any more compatible with media players.
2. Media players not supporting higher profiles are not necessarily
outdated (in fact this is simply an arrogant statement that
libavcodec can handle these particular features).
You could add that there are plenty of other ways to produce widely
incompatible files with ffmpeg, and these don't show any warnings.
What we really want to do here is defaulting to codec profiles that
have wide compatibility, such as main/high for h264. Also, if an
encoder does not accept certain pixfmts, we should automatically
convert them to a pixfmt the encoder can accept. But the existing
message certainly is not appropriate.
It also works for 2 specific encoders only. Extending it for other
cases would result in a lot of special cases, so this is not the
right place.
add a per-stream option for setting the encoder timebase.
the following values are allowed:
0 - for video, use 1/frame_rate, for audio use 1/sample_rate (this is
the default)
-1 - match the input timebase (when possible)
>0 - set the timebase to provided number
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Because write_packet() fakely writes packets to muxer by queueing
them when muxer hasn't been initialized, it should also increment
frame_number fakely.
This is required because code in do_streamcopy() rely on
frame_number.
Should fix Ticket6227
Reviewed-by: James Almer <jamrial@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
It is assigned from 64bit input in some branches and used with 64bit timestamps
This thus fixes a potential integer truncation
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Should fix tsan warnings in fate-fifo-muxer-h264/wav:
WARNING: ThreadSanitizer: data race (pid=26552)
Write of size 4 at 0x000001e0d7c0 by main thread:
#0 transcode_init src/ffmpeg.c:3761 (ffmpeg+0x00000050ca1c)
[..]
Previous read of size 4 at 0x000001e0d7c0 by thread T1:
#0 decode_interrupt_cb src/ffmpeg.c:460 (ffmpeg+0x0000004fde19)
The old "API" that signaled rotation as a metadata value has been
replaced by DISPLAYMATRIX side data quite a while ago.
There is no reason to make muxers/demuxers/API users support both. In
addition, the metadata API is dangerous, as user tags could "leak" into
it, creating unintended features or bugs.
ffmpeg CLI has to be updated to use the new API. In particular, we must
not allow to leak the "rotate" tag into the muxer. Some muxers will
catch this properly (like mov), but others (like mkv) can add it as
generic tag. Note applications, which use libavformat and assume the
old rotate API, will interpret such "rotate" user tags as rotate
metadata (which it is not), and incorrectly rotate the video.
The ffmpeg/ffplay tools drop the use of the old API for muxing and
demuxing, as all muxers/demuxers support the new API. This will mean
that the tools will not mistakenly interpret per-track "rotate" user
tags as rotate metadata. It will _not_ be treated as regression.
Unfortunately, hacks have been added, that allow the user to override
rotation by setting metadata explicitly, e.g. via
-metadata:s:v:0 rotate=0
See references to trac #4560. fate-filter-meta-4560-rotate0 tests this.
It's easier to adjust the hack for supporting it than arguing for its
removal, so ffmpeg CLI now explicitly catches this case, and essentially
replaces the "rotate" value with a display matrix side data. (It would
be easier for both user and implementation to create an explicit option
for rotation.)
When the code under FF_API_OLD_ROTATE_API is disabled, one FATE
reference file has to be updated (because "rotate" is not exported
anymore).
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
* 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>
