* commit '1d8a0c1b43e58332a3a15c67d4adc161713cade8':
movenc: Allow to request not to use edit lists
Conflicts:
libavformat/movenc.c
See: 537ef8bebf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '897d5c3a4296f3da80b8699d1487328ca2de8e55':
lavf: Print a warning if failed to avoid negative timestamps when requested
Conflicts:
libavformat/mux.c
See: ec6a5fc6cc
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When using the new first_trun flag instead of checking the track id,
we don't need to have a special case for the separate_moof flag
any longer.
This simplifies the complicated codepath ever so slightly.
Signed-off-by: Martin Storsjö <martin@martin.st>
In this case, shift tracks to start from zero instead (potentially
stretching the first sample in tracks that start later than the
first one).
Some software does not support edit lists at all, the adobe flash
player seems to be one of these. This results in AV sync errors when
edit lists are used to adjust AV sync.
Some players, such as QuickTime, don't respect the duration for
audio packets, so if an audio track starts later than the video
track and the first audio sample gets a duration longer than the
actual amount of data in it, the result will be out of sync.
Based on patches by Michael Niedermayer.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is the same logic as is invoked on AVFMT_TS_NEGATIVE,
but which can be enabled manually, or can be enabled
in muxers which only need it in certain conditions.
Also allow using the same mechanism to force streams to start
at 0.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '56dc46a1893251e74be1ad63e54fb38d754bb1fe':
riffenc: do not fall back on AVCodecContext.frame_size for MP3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '91e8d2eb1f7bf3af949008b106ec1ca037b88b0e':
lavf: use the format context strict_std_compliance instead of the codec one
Conflicts:
libavformat/mux.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2f3fadfbe3c6ad52fad5c614b6067c5401227959':
lavc,lavf: switch to the new vorbis parse API
Conflicts:
libavformat/oggparsevorbis.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5e80fb7ff226f136dbcf3fed00a2966bf8e9bd70':
lavc: add a public API for parsing vorbis packets.
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/version.h
libavcodec/vorbis_parser.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6896f95b2483e52e717e2c75a4fd24fcb0e14b67':
vorbis_parser: add an AV prefix to VorbisParseContext
Conflicts:
libavcodec/vorbis_parser.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The only parameters needed by the demuxers are the sample rate and sample
count, which can be trivially extracted manually, without resorting to
an avpriv function.
It will not be set unless the codec context is used as the encoding
context, which is discouraged. For MP2, av_get_audio_frame_duration()
will already set the frame size properly. For MP3, set the frame size
explicitly.
It is never an error if this method failed. If rt->live was
explicitly set to 0 (known to be a recorded file), print it
as a warning, otherwise print it as a debug message.
Based on a patch by Michael Niedermayer.
Signed-off-by: Martin Storsjö <martin@martin.st>
TDRL is what we used as a replacement of TYER, and, according to
http://id3.org/id3v2.4.0-changes :
TYER - Year
This frame is replaced by the TDRC frame, 'Recording time'
[F:4.2.5].
So change TDRL usages to TDRC.
Fixes ticket #3694
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This way if we by chance run into a valid PMT we have a more complete
set of streams, also do not reset streams in case we run into a worse
PMT
Fixes Ticket4046
alternatively Ticket4046 could be closed as invalid or wontfix as it contains
some PMTs which lack the 2 subtitle streams
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2f221b6a9365aa400061e16266f2d1242f7169f8':
movenc: Define the flag bits using shifts instead of as decimal numbers
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Avoid the creation of files which cannot be successfully decoded by
ffmpeg, for example generated with:
ffmpeg -f lavfi -i sine -af "aselect='not(between(t,100,500))',aresample=min_comp=0.001:min_hard_comp=0.100000" -acodec pcm_s16le -t 1000 -y out_audio.flv
