| Commit message (Collapse) | Author | Age |
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This helps in disentangling lavf and lavc too.
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Note that the old DNxHD UL is actually JPEG 2000 according to RP224.
Leaving it as-is for now.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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We cannot easily determine if an mpeg TS's packet size is DVHS, FEC
or so on, for that we need to expose the internal raw_packet_size
field.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Previously the default fragmentation method was only enabled
if writing an ISM file.
Signed-off-by: Martin Storsjö <martin@martin.st>
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@parblock is only supported in very recent Doxygen versions.
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Also drop support for building examples in library directories.
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Fixes fate-jv under valgrind which reports a different CRC for the last
frame from a partial read.
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After seeking fragments with an offset > 0 must be skipped to correctly
assemble packets.
Bug-Id: 43
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Bug-id: 43
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
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Hitting EOF during seeking is quite likely. Fixes use of uninitialized
data during fate-seek-lavf-asf.
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Remove some vague / incorrect statements and make the text more specific
and less confusing.
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Remove some vague / incorrect statements and make the text more specific
and less confusing.
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Libavformat should not make any assumptions about values returned from
other libraries. This assert is easily reachable by using a non-raw
codec id.
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Fixes memleak in fate-mxf-demux.
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Since 2007, the Xiph.org Foundation recommends that .ogg only be used
for Ogg Vorbis audio files.
Source: http://wiki.xiph.org/index.php/MIME_Types_and_File_Extensions
However we only do it if we have libvorbis available because the
built in vorbis encoder is not as good.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Does not contain cursor rendering yet.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Currently ff_interleave_packet_per_dts() waits until it gets a frame for
each stream before outputting packets in interleaved order.
Sparse streams (i.e. streams with much fewer packets than the other
streams, like subtitles or audio with DTX) tend to add up latency and in
specific cases end up allocating a large amount of memory.
Emit the top packet from the packet_buffer if it has a time delta
larger than a specified threshold.
Original report of the issue and initial proposed solution by
mus.svz@gmail.com.
Bug-id: 31
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Those streams should never get any packets by definition.
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Reject packets for non-existing or attachment streams.
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And export the information in field_order.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Default to 1/25.
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Due to what looks like an off-by-one error 60 * 12 * 1001 / 12 * 1001
is not tested as standard frame rate in avformat_find_stream_info().
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Also adjust some comment wording / spelling and reorder header #includes.
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This fixes playback of mp3 streams in rtp/asf. This used to work
until c6f1dc8, but mostly by coincidence.
Signed-off-by: Martin Storsjö <martin@martin.st>
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The normal differential timestamps can't handle negative
differences, thus send a full packet header with an absolute
timestamp in these cases.
Signed-off-by: Martin Storsjö <martin@martin.st>
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If the url ends with .flv, we stripped it but didn't initialize
rt->playpath, doing av_strlcat on an uninitialized buffer.
Signed-off-by: Martin Storsjö <martin@martin.st>
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The track duration is often not reliable or is not the duration
represented by the number of frames. In those cases, avg_frame_rate
was reported incorrectly. Removing this code falls back to the
default calculation in avformat_find_stream_info().
This is a partial revert of commit c3aeaa540.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Reviewed-by: Stephen Hutchinson <qyot27@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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It could probably also be considered an error if the pointer isn't
null at this point, but then we might risk rejecting some
slightly broken files that we might have handled so far.
Sample-Id: 00000496-google
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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These arrays are normally freed at the end of mov_read_trak,
but make sure they're freed in case mov_read_trak returned
early (due to errors) or in case the atoms that allocate arrays
are encountered at some other point than within a trak (which
we don't have checks against).
Sample-Id: 00000496-google
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Such files have IndexTableSegments which when parsed cover EditUnit
ranges like this:
[0,1)
[249,250)
[249,377)
[0,249)
where each interval is
[IndexStartPosition, IndexStartPosition + IndexDuration)
This would be reduced to a sparse index like:
[0,1), [249,250)
instead of the full range:
[0,249), [249,377)
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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av_reallocp_array does the check already.
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Without this fix the last sample was missing from the packet.
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Overly large files are to be expected.
Reported-by: Jean-Baptiste Kempf <jb@videolan.org>
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Needed for AVC-intra
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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