According to the HEIF specification (ISO/IEC 23008-12) Section
7.5.3.1, tracks with handler_type 'auxv' must contain a 'auxi' box
in its SampleEntry to notify the nature of the auxiliary track to the
decoder.
The content is the same as the 'auxC' box. So parameterize and re-use
the existing function.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: James Zern <jzern@google.com>
Fixes: signed integer overflow: 538976288 * 4 cannot be represented in type 'int'
Fixes: 50993/clusterfuzz-testcase-minimized-ffmpeg_dem_ICO_fuzzer-6690068904935424
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 3 * -2147483648 cannot be represented in type 'int'
Fixes: 50993/clusterfuzz-testcase-minimized-ffmpeg_dem_AIFF_fuzzer-6668935979728896
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 3 * -2147483648 cannot be represented in type 'int'
Fixes: 50993/clusterfuzz-testcase-minimized-ffmpeg_dem_AIFF_fuzzer-6668935979728896
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 9223372036854775807 - -2146905566 cannot be represented in type 'long'
Fixes: 50993/clusterfuzz-testcase-minimized-ffmpeg_dem_MXF_fuzzer-6570996594769920
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In case a SupplementalProperty node exists in an adaptationset,
it is searched for a "schemeIdUri" property via xmlGetProp().
Whatever xmlGetProp() returns is then compared via av_strcasecmp()
to a string literal. xmlGetProp() can return NULL, namely in case
no "schemeIdUri" exists and (given that this string is allocated)
presumably also on allocation failure. No check for NULL is done,
so this may crash.
Furthermore, the string returned by xmlGetProp() needs to be freed
with xmlFree(), but this is not done either.
This commit fixes both of these issues; they existed since this code
has been added in 10d008f0fd.
This has been found while investigating ticket #9697. The continuous
leaks might very well be the reason behind the observed slowdown.
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
When determining whether a packet should be decrypted,
should use the stsd_id of the fragment where the current packet is located.
Reviewed-by: Zhao Zhili <zhilizhao@tencent.com>
Signed-off-by: Wang Yaqiang <wangyaqiang03@kuaishou.com>
Clang's static analyzer complains that leaving the variable
uninitialized could lead to a code path where the uninitialized value is
written to at the end of this function.
This patch simply zero-initializes that variable to avoid that.
Signed-off-by: Will Cassella <cassew@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The MXF demuxer does not currently set AVStream::avg_frame_rate and ::r_frame_rate
when J2K essence is wrapped according to SMPTE ST 422.
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Just because we try to put multiple units of block_align bytes
(the atomic units for APTX and APTX HD) into one packet
does not mean that packets with fewer units than the
one we wanted are corrupt; only those packets that are not
a multiple of block_align are.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This field was misunderstood: It gives the number of samples
in a packet, not the number of bytes. Its usage was wrong for APTX HD.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
av_strlcpy() returns the length of the src string to enable
the caller to check for truncation. It is currently used in
the following way in dump_metadata(): Every metadata value
is searched for \b, \n, \v, \f, \r and then the data up to
the first of these characters found is copied to a small
temporary buffer via av_strlcpy() (but of course not more
than fits into said buffer) and then printed; all characters up
to the character found earlier are then treated as consumed.
But this is bad performance-wise if the while string is big
and contains many of these characters, because av_strlcpy()
will unnecessarily calculate the length of the whole remaining string.
(dump_metadata() actually ignored the return value of av_strlcpy().)
Fix this by not copying the data to a temporary buffer at all.
Instead just use %.*s to bound the number of characters output.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes: memleak
Fixes: 50703/clusterfuzz-testcase-minimized-ffmpeg_dem_HLS_fuzzer-6399058578636800
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This duration is equal to the longest duration in all track's tkhd atoms, which
may be comprised of the sum of all edit lists in each track. Empty edit lists
in tracks represent start_time, and the actual media duration is stored in the
mdhd atom.
This change lets the generic demux code derive the longest track duration taken
from mdhd atoms, so the correct duration and start_time combination will be
reported.
Should fix ticket #9775.
Reviewed-by: zhilizhao(赵志立) <quinkblack@foxmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The field is not specific to Opus.
The mp2fixed encoder signals initial_padding and is used
by both the matroska-encoding-delay test as well as
the lavf-mkv tests which necessitated several FATE ref changes.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Matroska requires pts to be >= 0 with a slight exception:
It has a mechanism to deal with codec delay, i.e. with
the data added at the beginning that does not correspond
to actual input data and should be discarded by the player.
Only the audio actually intended to be output needs to have
a timestamp >= 0.
In order to avoid unnecessary timestamp shifting, this patch
allows muxers to inform the shifting code about this so that
it can take it into account.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Matroska generally requires timestamps to be nonnegative, but
there is an exception: Data that corresponds to encoder delay
and is not supposed to be output anyway can have a negative
timestamp. This is achieved by using the CodecDelay header
field: The demuxer has to subtract this value from the raw
(nonnegative) timestamps of the corresponding track.
Therefore the muxer has to add this value first to write
this raw timestamp.
Support for writing CodecDelay has been added in FFmpeg commit
d92b1b1bab and in Libav commit
a1aa37dd0b. The former simply
wrote the header field and did not apply any timestamp offsets,
leading to desynchronisation (if one uses multiple tracks).
The latter applied it at two places, but not at the one where
it actually matters, namely in mkv_write_block(), leading to
the same desynchronisation as with the former commit. It furthermore
used the wrong stream timebase to convert the delay to the
stream's timebase, as the conversion used the timebase from
before avpriv_set_pts_info().
When the latter was merged in 82e4f39883,
it was only done in a deactivated state that still did not
offset the timestamps when muxing due to "assertion failures
and av sync errors". a1aa37dd0b
made it definitely more likely to run into assertion failures
(namely if the relative block timestamp doesn't fit into an int16_t).
Yet all of the above issues have been fixed (in commits
962d631573,
5d3953a5dc and
4ebeab15b0. This commit therefore
enables applying CodecDelay, fixing ticket #7182.
There is just one slight regression from this: If one has input
with encoder delay where the first timestamp is negative, but
the pts of the part of the data that is actually intended to be
output is nonnegative, then the timestamps will currently by default
be shifted to make them nonnegative before they reach the muxer;
the muxer will then ensure that the shifted timestamps are retained.
Before this commit, the muxer did not ensure this; instead the
timestamps that the demuxer will output were shifted and
if the first timestamp of the actually intended output was zero
before shifting, then this unintentional shift just cancels
the shift performed before the packet reached the muxer.
(But notice that this only applies if all the tracks use the same
CodecDelay, or the relative sync between tracks will be impaired.)
This happens in the matroska-opus-remux and matroska-ogg-opus-remux
FATE tests. Future commits will forward the information that
the Matroska muxer has a limited capability to handle negative
timestamps so that the shifting in libavformat can take advantage
of it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Opus can be decoded to multiple samplerates (namely 48kHz, 24KHz,
16Khz, 12 KHz and 8Khz); libopus as well as our encoder wrapper
support these sample rates. The OpusHead contains a field for
this original samplerate. Yet the pre-skip (and the granule-position
in the Ogg-Opus mapping in general) are always in the 48KHz clock,
irrespective of the original sample rate.
Before commit c3c22bee63, our libopus
encoder was buggy: It did not account for the fact that the pre-skip
field is always according to a 48kHz clock and wrote a too small
value in case one uses the encoder with a sample rate other than 48kHz;
this discrepancy between CodecDelay and OpusHead led to Firefox
rejecting such streams.
In order to account for that, said commit made the muxer always use
48kHz instead of the actual sample rate to convert the initial_padding
(in samples in the stream's sample rate) to ns. This meant that both
fields are now off by the same factor, so Firefox was happy.
Then commit f4bdeddc3c fixed the issue
in libopusenc; so the OpusHead is correct, but the CodecDelay is
still off*. This commit fixes this by effectively reverting
c3c22bee63.
*: Firefox seems to no longer abort when CodecDelay and OpusHead
are off.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is possible for the trailing padding to be zero, namely
e.g. if the AV_PKT_DATA_SKIP_SAMPLES side data is used
for leading padding. Matroska supports this (use a negative
DiscardPadding), but players do not; at least Firefox refuses
to play such a file. So for now only write DiscardPadding
if it is trailing padding and nonzero.
The fate-matroska-ogg-opus-remux was affected by this.
(I wish CodecDelay would not exist and DiscardPadding would
be used to instead trim the codec delay away (with the Block
timestamp corresponding to the time at which the actually
output audio is output).)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Demuxers are not allowed to do this and few callers, if any, will handle
this correctly. Send the AV_SIDE_DATA_PARAM_CHANGE_SAMPLE_RATE side data
instead.
Demuxers are not supposed to update AVCodecParameters after the stream
was seen by the caller. This value is not important enough to support
dynamic updates for.
The mov demuxer only returns DV audio, video packets are discarded.
It first reads the data to be parsed into a packet. Then both this
packet and the pointer to its data are passed together to
avpriv_dv_produce_packet(), which parses the data and partially
overwrites the packet. This is confusing and potentially dangerous, so
just pass NULL and avoid pointless packet modification.
Initialized to 1:1, but if the script sets these properties, it
will be set to those instead (0:0 disables it, apparently).
Signed-off-by: Stephen Hutchinson <qyot27@gmail.com>
It makes no sense here, as flac_parse_block_header()
is not even supposed to advance the caller's pointer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The HEVC code currently uses an array of arrays of NALUs; one such array
contains all the SPS NALUs, one all PPS NALUs etc. The array of arrays
is grown dynamically via av_reallocp_array(), but given that the latter
function automatically frees its buffer upon reallocation error,
it may only be used with PODs, which this case is not. Even worse:
While the pointer to the arrays is reset, the counter for the number
of arrays is not, leading to a segfault in hvcc_close().
Fix this by avoiding the allocations of the array of arrays altogether.
This is easily possible because their number is bounded (by five).
Furthermore, as a byproduct we can ensure that the code always
produces the recommended ordering of VPS-SPS-PPS-SEI (which was
not guaranteed before).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Vignesh Venkatasubramanian
Michael Niedermayer
Andreas Rheinhardt
Wang Yaqiang
Will Cassella
Pierre-Anthony Lemieux
Paul B Mahol
James Almer
Anton Khirnov
Stephen Hutchinson