According to the DASH spec, Representation IDs should be unique
across all adaptation sets. Fixing that and updating the fate
reference file to reflect this change.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Using 100-continue ffmpeg will only send data if the server confirms it,
so if there is an error with auth or mounpoint, this allows that it is
properly reported to the user. Else ffmpeg sends data and just quits at
some point without an error message.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
use a default (audio/mpeg for historical reason) if none. Required since Icecast 2.4.1
Not using AVOption default because this breaks content-type warnings (needs to
detect if no type was set by the user)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This allows selecting if the demuxer should consider all streams to be
found after the first PMT and add further streams during decoding or if it rather
should scan all that are within the analyze-duration and other limits
Fixes Ticket3762
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8cb7b7b461b52898765b38e3eff68c0ce88347f3':
movenc: Avoid leaking locally allocated data when returning on errors
Conflicts:
libavformat/movenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids a potential crash if writing a fragmented psp mp4
(which probably is only a hypothetical scenario).
Signed-off-by: Martin Storsjö <martin@martin.st>
In matroska_read_seek(), |tracks| is assigned at the begining of the function.
However, functions like matroska_parse_cues() could reallocate the tracks so
that |tracks| can get invalidated.
This CL assigns |tracks| only before we use it so that it won't be invalidated.
BUG=427266
TEST=Test case in associated bug passes now.
Change-Id: I9c7065fe8f4311ca846076281df2282d190ed344
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
Previously we wrote decoding timestamps here, while the specs
say it should be presentation timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
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.
