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Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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This causes regressions in end to end timestamps with mp3s and ffmpeg.
The revert is to avoid this regression in the 4.3 release
See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted.
This reverts commit 460132c9980f8a1f501a1f69477bca49e1641233.
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Fixes: memleak
Fixes: 23142/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5932860820422656
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fixes: NULL pointer dereference
Fixes: 22604/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5667739074297856.fuzz
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fixes: out of array access
Fixes: 22520/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5100297658826752
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fixes: regression since e983197cbc93420b67aa7e811be47d7278c2c8a2
Fixes: out of array read
Fixes: 22185/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5662069073641472
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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everywhere
Fixes regression since 9ad47762c17d2c6d06595aa17b88112baa91b72c
Fixes: out of array access
Fixes: 22172/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5658535590625280
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fixes: out of array access
Fixes: 22686/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5121369624018944
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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decoder's codec.
Prevent codecpar->codec_id from getting out of sync with the codec instantiated for probing.
Signed-off-by: Samuel Foss <sfoss@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fix trac ticket #8705.
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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Previously we always set STREAM_TYPE_PRIVATE_DATA, and that remains
the default value.
Signed-off-by: Marton Balint <cus@passwd.hu>
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Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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e,g: the command:
ffprobe -show_format -i fate-suite/aac/foo.aac -loglevel 99 will
dump the trace message as follow when start_time is AV_NOPTS_VALUE
[aac @ 0x55bf8e1f3dc0] stream 0: start_time: -326791809695.818 duration: 2.174
[aac @ 0x55bf8e1f3dc0] format: start_time: -9223372036854.775 duration: 2.174 bitrate=120 kb/s
after this fix, will dump the start_time with "NOPTS".
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
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Avoids overflow from fuzzed skip_samples values.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Up until now, the HLS muxer uses av_strtok() to split an input string
controlling parameters of the VariantStreams and then duplicates
parts of this string containing parameters such as the language or the
name of the VariantStream. But these parts are proper zero-terminated
strings of their own that are never modified lateron, so one can simply
use the substring as-is without creating a copy. This commit implements
this.
The same also happened for the string controlling the closed caption
groups.
Furthermore, add const to indicate that the pointers to these substrings
are not used to modify them and also to indicate that these strings are
not allocated on their own.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Up until now, the HLS muxer duplicated a string for every VariantStream,
although neither the original nor the copies are ever modified. So use
the original directly and stop copying.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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commit 32aeba12755 missed coding style fix.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
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raw_packet_size
The old resync logic had some bugs, for example the packet size could stuck
into 192 bytes, because pos47_full was not updated for every packet, and for
unseekable inputs the resync logic simply skipped some 0x47 sync bytes,
therefore the calculated distance between sync bytes was a multiple of 188
bytes.
AVIO only buffers a single packet (for UDP/mpegts, that usually means 1316
bytes), so for every ten consecutive 188-byte MPEGTS packets there was always a
seek failure, and that caused the old code to not find the 188 byte pattern
across 10 consecutive packets.
This patch changes the custom logic to the one which is used when probing to
determine the packet size. This was already proposed as a FIXME a long time
ago...
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Uses ff_get_wav_header() in riffdec.c
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Merge the short lines after the last commit
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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7546ac2fee4 made it so that the start_time for mp3 files is
adjusted for skip_samples. However, this appears incorrect because
subsequent packet timestamps are not adjusted and skip_samples are
applied by deleting data from a packet without changing the timestamp.
E.g., we are told the start_time is ~25ms and we get a packet with a
timestamp of 0 that has had the skip_samples discarded from it. As such
rendering engines may incorrectly discard everything prior to the
25ms thinking that is where playback should officially start. Since the
samples were deleted without adjusting timestamps though, the true
start_time is still 0.
Other formats like MP4 with edit lists will adjust both the start
time and the timestamps of subsequent packets to avoid this issue.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Fixes: memleak
Fixes: 19949/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5743636058210304
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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It avoids leaving dangling pointers behind in memory.
Also remove redundant checks for whether the URLContext to be closed is
already NULL.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Some flac muxers write truncated metadata picture size if the picture
data do not fit in 24 bits. Detect this by truncting the size found inside
the picture block and if it matches the block size use it and read rest
of picture data.
This workaround is only for flac files and not ogg files with flac
METADATA_BLOCK_PICTURE comments and it can be disabled with strict level
above normal. Currently there is a 500MB limit on truncate size to protect
from large memory allocations.
The truncation bug in lavf flacenc was fixed in e447a4d112bcfee10126c54eb4481fa8712957c8
but based on existing broken files other unknown flac muxers seems to truncate also.
Before the fix a broken flac file for reproduction could be generated with:
ffmpeg -f lavfi -i sine -f lavfi -i color=red:size=2400x2400 -map 0:0 -map 1:0 -c:v:0 bmp -disposition:1 attached_pic -t 1 test.flac
Fixes ticket 6333
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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ff_id3v2_parse_apic/chapters/priv/priv_dict all had a parameter
extra_meta of type ID3v2ExtraMeta ** as if the functions wanted to make
*extra_meta point to something else. But they don't, so just use an
ID3v2ExtraMeta *.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Up until now, the ID3v2ExtraMeta structure (which is used when parsing
ID3v2 tags containing attached pictures, chapters etc.) contained a
pointer to separately allocated data that depended on the type of the
tag. Yet the difference of the sizes of the largest and the smallest of
these structures is fairly small, so that it is better to simply include
a union of all the possible types of tag-dependent structures in
ID3v2ExtraMeta. This commit implements this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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If the write_id3v2 option is set, the aiff muxer would write id3v2 tags
if there is global metadata or if there are attached pics to write.
Chapters are ignored in this check that precedes writing id3v2 tags.
Yet 47ac344970f1d6ef149c4b8a883b68cdb15112c2 added support for writing
chapters as id3v2 tags, so one should check for the existence of chapters,
too; otherwise the chapters would only be written in case there is
global metadata or an attached pic.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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The Xiph foundation never standardized either Daala nor its mapping in Ogg,
and all files that were created are undecodable without knowledge of the
git hash.
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The description of AVOutputFormat.init contains the statement that "this
method must not write output". Due to this, the webm_chunk muxer defers
opening the AVIOContext for the child muxer until avformat_write_header(),
i.e. there is no AVIOContext when the sub-muxer's avformat_init_output()
is called. But this violates the documentation of said function which
requires the AVFormatContext to have an already opened AVIOContext.
This commit fixes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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by storing s->streams[i] in a pointer instead of constantly using
s->streams[i]->...
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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