They are optional and just define a channel layout, which may also be defined
by the underlying codec.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 95fe0658d7)
Needed to keep the process going if some issue was found while parsing these boxes.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit fd1c8fa0e6)
celt_header() reads a uint32 `extra_headers` field from the CELT identification
header and stores `1 + extra_headers` into the signed int extra_headers_left.
With extra_headers = 0x7FFFFFFE this becomes INT_MAX and the OGG parser
consumes every subsequent page as a CELT "extra header" without ever reaching
audio data, hanging on any streaming input. A value of 0xFFFFFFFE wraps the
signed addition negative, with the same family of consequences.
Reject any extra_headers count above a small fixed cap (16, well above any
real CELT-over-Ogg stream).
Verified with the audit PoC (a crafted file plus an infinite-page FIFO):
without the patch, ffmpeg consumes pages forever; with the patch it logs
"Too many CELT extra headers (...)" and exits in ~70 ms with
AVERROR_INVALIDDATA.
Reported by Franciszek Kalinowski (isec.pl / striga.ai) and Bartosz Smigielski.
(cherry picked from commit 87439ed619)
You may look and think `AVFMT_FLAG_CUSTOM_IO` check is enough, but this
is not what it seems. This flag means that user provided custom
AVIOContext, before creating AVFormatContext and it should not be
closed. However nested sub-demuxers may still open an temporary io, and
those have to be closed and use correct io_close2 function.
You can see 0dcac9c3f0 and
ef01061225 where this flag is cleared for
nested opens to avoid leaking those.
lavf micro version bumped so API users can know if it is safe to use
custom io.
(cherry picked from commit dc74fe70b2)
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
io_open_default() will call internal impl if needed, don't call it
directly.
(cherry picked from commit 06ef9a74ea)
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
io_open_default() will call internal impl if needed, don't call it
directly.
(cherry picked from commit 4cf687b3b1)
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
io_open_default() will call internal impl if needed, don't call it
directly.
(cherry picked from commit fbc4003642)
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
Fixes ticket #22890.
The ff_hls_senc_parse_audio_setup_info function passes setup_data to
parsers like avpriv_ac3_parse_header and init_get_bits8 which require
the buffer to be padded with AV_INPUT_BUFFER_PADDING_SIZE bytes at the end.
(cherry picked from commit 71d5fa8d4d)
Signed-off-by: Marvin Scholz <epirat07@gmail.com>
Given the entire iamf struct is inside a single Track, if the first iamf stream
(which is the one sharing the index and id from the Track) was to be disabled,
then packets from every iamf stream would be discarded.
Fix this by actually going through the entire iamf Sample and discarding those
from the disabled streams only.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 16ee3d8d99)
If we were to add the highest id of a non iamf stream as offset to iamf stream
ids, and one of the latter was 0, then an id overlap would ocurr.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 51aef95ba1)
Using audio_substream_id for AVStream ids is not ideal give that in containers
like mp4, the IAMF structure is opaque to the outside and other streams may
share such id values.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit a8f2374507)
Not entirely sure if it should instead use some entirely different
approach here, given that images exceeding 2GB don't seem that crazy
to me, but so far processing such images results in a heap overflow,
since the size addition overflows and a much too small packet is
allocated and its size never checked again when writing into it.
Fixes #YWH-PGM40646-32
(cherry picked from commit f6a95c7eb7)
- proper pts for packets. leaving it blank leaves it up for guessing,
but the guess doesn't take seeking into account, causing weirdness.
- clamp to 0 when seeking to negative ts. libopenmpt docs are unclear on
this but not doing this causes an immediate EOF when seeking backwards
to the beginning in mpv.
- only set song duration and packet pts when they are non-negative and
in int64 range. NaNs count as out of range. this isn't a fix for any
specific issue but might be helpful still, and shouldn't break
anything.
(cherry picked from commit ecef5f9e1f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Apparently files with milliseconds exist in the wild. And since it cost
nothing to support arbitrary number of digits, extend format to support
that.
Depending on number of digits, the time base of fractional part is
changing. Most LRCs use 2 digits and centiseconds base, but subs with 3
digits and miliseconds exist too.
Set internal time base to AV_TIME_BASE, which in parcitice allows to
hold microseconds with 6 digits. Totally artificial, but who knows maybe
someone wants that.
Fixes: #11677
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
(cherry picked from commit bc3cc0a6af)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: writing 1 byte over the end of the array
Fixes: BIGSLEEP-433502298/test.xml
Found-by: Google Big Sleep
A prettier solution is welcome!
A testcase exists only for the baseurl case
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ce0a655f85)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows the user to set only the one that is needed to ALL or a
specific "wrong" extension like html
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f99f223eb1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes part of Ticket11435
Fixes: Elisa Viihde (Finnish online recording service)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 68644994fd)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(setting to 100 as a reasonable compromise)
The change has caused regressions for many users and consumers.
Playlist reloads only happen when a playlist doesn't indicate that it
has ended (via #EXT-X-ENDLIST), which means that the addition of future
segments is still expected.
It is well possible that an HLS server is temporarily unable to serve
further segments but resumes after some time, either indicating a
discontinuity or even by fully catching up.
With a segment length of 3s, a max_reload value of 1000 corresponds to
a duration of 50 minutes which appears to be a reasonable default.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ace9f03a6c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
As per section 3.11.1 of the IAMF spec, the sample rate used in Codec Config
for Opus shall be 48kHz, regardless of the original sample rate used during
encoding.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 76049d1c45)
When subblock durations are constant, the last block may be smaller and the
value needs to be calculated.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit fb5e8ea971)
Section 3.6.1 of the IAMF spec states "When constant_subblock_duration is equal to 0, the summation of all
subblock_duration in this parameter block SHALL be equal to duration.".
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit d38fc25519)
