This patch makes it possible to dinamically close the current segment
and step to the next one by introducing command handling capabilities
into the filter. This new feature is very usefull when working with
real-time sources or live streams as source. Combinig usage with zmqsend
tool you can interactively end the current segment and step to next one.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
Fixes: 6154/clusterfuzz-testcase-minimized-5762231061970944
Fixes: runtime error: shift exponent 63 is too large for 32-bit type 'int'
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
External headers are no longer welcome in the ffmpeg codebase because they
increase the maintenance burden. However, in the NVidia case the vanilla
headers need some modifications to be usable in ffmpeg therefore we still
provide them, but in a separate repository.
The external headers can be found at
https://git.videolan.org/?p=ffmpeg/nv-codec-headers.git
Fate-source is updated because of the deleted files, and dynlink_loader.h
license headers were updated with the standard FFmpeg headers.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
Fixes: Timeout
Fixes: 6266/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RSCC_fuzzer-5692431816196096
Its not known if nb_tiles is allowed so it is not treated as an error
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This only affected demuxers that didn't return reference counted packets.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
If configure fails before config.fate is generated, the report file misses
some values and gets discarded by the FATE server. In these cases, print
those values as "failed" along with the failing configure command line.
ffmpeg -h display "max_error_rate" option help information have
been cut off, the root cause is used a wrong initial order.
Signed-off-by: Jun Zhao <jun.zhao@intel.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
documents delete_filler option for bsf h264_metadata.
Signed-off-by: Jun Zhao <jun.zhao@intel.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Prevents use of uninitialized values.
Fixes ticket #7038.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
The supplementary audio descriptor is defined in ETSI EN 300 468 and
provides more details regarding accessibility audio tracks, especially
the normative annex J contains a detailed description of its use.
Its language code (if present) overrides the language code of an also
present ISO 639 language descriptor.
Note that this also changes the priority of multiple descriptors with
language from "the last descriptor with language within the ES loop" to
"specific descriptor over general ISO 639 descriptor".
Signed-off-by: Aman Gupta <aman@tmm1.net>
Currently using HLS_TEMP together with HLS_SECOND_LEVEL_SEGMENT_DURATION
or HLS_SECOND_LEVEL_SEGMENT_SIZE gives error at end of each segment
writing and the final segment file names do not contain the desired
data. This patch fixes this bug by delaying the initilization of
original segment filename after actual temp file renaming will skip the
interfering.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
we found some very old videos which suffered from
corruption after 9e6a242755, but were fine
before.
These had "End of AC stream reached in vp6_parse_coeff" warnings in logs.
These also had flv Packet mismatch warnings.
Adding FlixEngine to the list of flv muxers which produce broken packet
sizes fixes this corruption.
FlixEngine is very old and not maintained or available anymore (since
2010), so we won't need to worry about newer versions fixing the issue.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The PicStruct is required by MediaSDK, so give a default value.
hwupload does not work without this.
Signed-off-by: Ruiling Song <ruiling.song@intel.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This will result in poor quality audio for SSR streams, but they
will at least demux and decode without error; partially fixing
ticket #1693.
This pulls in the decode_gain_control() function from the
ffmpeg summer-of-code repo (original author Maxim Gavrilov) at
svn://svn.mplayerhq.hu/soc/aac/aac.c with some minor modifications
and adds AOT_AAC_SSR to decode_audio_specific_config_gb().
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Co-authored-by: Maxim Gavrilov <maxim.gavrilov@gmail.com>
