| Commit message (Collapse) | Author | Age | Files | Lines |
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CC: libav-stable@libav.org
Bug-Id: CID 1087094
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CC: libav-stable@libav.org
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It is av_opt_set now.
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Add SPS_EXT, SEI, and subset SPS to codes that are skipped during split.
These codes can come before the PPS and results in incomplete extradata.
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Bug-Id: CID 1087088
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Fixes the comparison against constant value 0x7fffffff.
CC: libav-stable@libav.org
Bug-Id: CID 1198260
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CC: libav-stable@libav.org
Bug-Id: CID 1238839
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Use av_mallocz_array instead of iterating and check the returned memory.
Check returned memory and cleanly exit in case of error during the loop.
Avoid a null pointer dereference for invalid data.
CC: libav-stable@libav.org
Bug-Id: CID 29575
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Fix a memory leak too.
CC: libav-stable@libav.org
Bug-Id: CID 732230
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Prevents shifts with undefined behavior, as no syntax element has a valid
value greater than 1 << 31.
Bug-Id: CID 1206635
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CC: libav-stable@libav.org
Bug-Id: CID 1244202
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data_size is always >= 0 in this case.
CC: libav-stable@libav.org
Bug-Id: CID 1238788
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CC: libav-stable@libav.org
Bug-Id: CID 991852
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CC: libav-stable@libav.org
Bug-Id: CID 1047235
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CC: libav-stable@libav.org
Bug-Id: CID 1238789
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CC: libav-stable@libav.org
Bug-Id: CID 1238786
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CC: libav-stable@libav.org
Bug-Id: CID 1047236
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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vorbis_parser.o is built unconditionally since 5e80fb7ff, and the
unconditionally built parts of it depend on xiph.o.
This fixes builds with --disable-everything.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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The latest fdk-aac code drop (from android 5.0) changed the channel
layout enums (changing the value of existing enum constants), and
renamed the option for downmixing.
The failsafe comparison between ctype and FF_ARRAY_ELEMS(channel_counts)
can trigger warnings (-Wtautological-constant-out-of-range-compare)
when building with the old FDK AAC releases, where it can't be
out of range with the enum values used there.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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av_log usually resets it.
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Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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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>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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QuickTime does support files with an empty initial movie
these days.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Previously we wrote decoding timestamps here, while the specs
say it should be presentation timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
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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>
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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>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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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>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Add a comment to explain the code.
CC: libav-stable@libav.org
Bug-Id: CID 1194388
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CC: libav-stable@libav.org
Bug-Id: CID 1135767
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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CC: libav-stable@libav.org
Bug-Id: CID 717453 / CID 717454 / CID 717915
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Makes coverity less confused and the code more readable.
CC: libav-stable@libav.org
Bug-Id: CID 751744
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CC: libav-stable@libav.org
Bug-Id: CID 608053
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Remove stray semicolon and share codepaths.
CC: libav-stable@libav.org
Bug-Id: CID 991857
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It is not supposed to be set by demuxers.
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The reasoning is the same as for
0097cbea695e534fce39958ccd103af2fbf65831.
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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.
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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.
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It is required by (at least) the ogg demuxer.
Mark the current semi-public apriv API for removal.
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This is done in preparation for making it public.
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Currently, the API takes an external AVCodecContext, which is used only
for extradata and logging. This change will allow to it to work without
an AVCodecContext in the following commits.
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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>
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