The old approach used some highly complex delta computation math and
output-delaying.
I do not remember what the initial reasoning behind that was, but given
that we can just offset the dts by the amount of bframes, it seems wholy
unnecessary.

This leaves open an issue with VFR content, for which some more complex
logic might be needed.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>

Due to a typo, it was impossible to write 0.595 / -4.5 dB
of ltrt_cmixlev, ltrt_surmixlev, loro_cmixlev, loro_surmixlev.
Without any error 0.841 / -1.5 dB was written to file.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Fixes: out of array write
Fixes: Regression since f619e1ecReviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Rav1e currently uses the time base given to it only for ratecontrol... where
the inverse is taken and used as a framerate. So, do what we do in other wrappers
and use the framerate if we can.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Fixes ticket #7303
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>

Sequence numbers of segments should be unique, if an encoder is using shorter
than 1 second segments and it is restarted, then future segments will be using
already used sequence numbers if initial sequence number is based on the number
of seconds since epoch and not microseconds.
Signed-off-by: Marton Balint <cus@passwd.hu>

Fixes problems when non-rational options were set using rational expressions,
causing rounding errors and the option range limits not to be enforced
properly.

ffmpeg -f lavfi -i "sine=r=96000/2"

This caused an assertion failure with assert level 2.
Signed-off-by: Marton Balint <cus@passwd.hu>

Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Reindentation as well as marking several variables used for demuxing
RealAudio as const to clearly see that they don't change during
demuxing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

The Matroska demuxer has three functions for creating packets out of
the data read: One for certain RealAudio codecs (ATRAC3, cook, sipr,
RealAudio 28.8), one for WebVTT (actually, the WebM flavour of it) and
one for all the others. Only the last function supported Matroska's
ContentCompression (e.g. it reversed zlib compression or added the
removed headers to the packets). But in Matroska, all tracks are allowed
to be compressed. This commit adds support for this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

There is no need to recheck this for every frame.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

Matroska is built around the principle that a reader does not need to
understand everything in a file in order to be able to make use of it;
it just needs to ignore the data it doesn't know about.

Our demuxer typically follows this principle, but there is one important
instance where it does not: A Block belonging to a TrackEntry with no
associated stream is treated as invalid data (i.e. the demuxer will try
to resync to the next level 1 element because it takes this as a sign
that it has lost sync). Given that we do not create streams if we don't
know or don't support the type of the TrackEntry, this impairs this
demuxer's forward compability.

Furthermore, ignoring Blocks belonging to a TrackEntry without
corresponding stream can (in future commits) also be used to ignore
TrackEntries with obviously bogus entries without affecting the other
TrackEntries (by not creating a stream for said TrackEntry).

Finally, given that matroska_find_track_by_num() already emits its own
error message in case there is no TrackEntry with a given TrackNumber,
the error message (with level AV_LOG_INFO) for this can be removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

A Block (meaning both a Block in a BlockGroup as well as a SimpleBlock)
must have at least three bytes after the field containing the encoded
TrackNumber. So if there are <= 3 bytes, the Matroska demuxer would
skip this block, believing it to be an empty, but valid Block.

This might discard valid nonempty Blocks, namely if the track uses header
stripping. And certain definitely spec-incompliant Blocks don't raise
errors: Those with two or less bytes left after the encoded TrackNumber
and those with three bytes left, but with flags indicating that the Block
uses lacing as then there has to be further data describing the lacing.

Furthermore, zero-sized packets were still possible because only the
size of the last entry of a lace was checked.

This commit fixes this. All spec-compliant Blocks that contain data
(even if side data only) are now returned to the caller; spec-compliant
Blocks that don't contain anything are not returned.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

Some conditions which don't change and which can therefore be checked
in read_header() were instead rechecked upon parsing each block. This
has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

The Matroska demuxer splits every sequence of h Matroska Blocks into
h * w / cfs packets of size cfs; here h (sub_packet_h), w (frame_size)
and cfs (coded_framesize) are parameters from the track's CodecPrivate.

It does this by splitting the Block's data in h/2 pieces of size cfs each
and putting them into a buffer at offset m * 2 * w + n * cfs where
m (range 0..(h/2 - 1)) indicates the index of the current piece in the
current Block and n (range 0..(h - 1)) is the index of the current Block
in the current sequence of Blocks. The data in this buffer is then used
for the output packets.

The problem is that there is currently no check to actually guarantee
that no uninitialized data will be output. One instance where this is
trivially so is if h == 1; another is if cfs * h is so small that the
input pieces do not cover everything that is output. In order to
preclude this, rmdec.c checks for h * cfs == 2 * w and h >= 2. The
former requirement certainly makes much sense, as it means that for
every given m the input pieces (corresponding to the h different values
of n) form a nonoverlapping partition of the two adjacent frames of size w
corresponding to m. But precluding h == 1 is not enough, other odd
values can cause problems, too. That is because the assumption behind
the code is that h frames of size w contain data to be output, although
the real number is h/2 * 2. E.g. for h = 3, cfs = 2 and w = 3 the
current code would output four (== h * w / cfs) packets. although only
data for three (== h/2 * h) packets has been read.

(Notice that if h * cfs == 2 * w, h being even is equivalent to
cfs dividing w; the latter condition also seems very reasonable:
It means that the subframes are a partition of the frames.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

RealAudio 28.8 (like other RealAudio codecs) uses a special demuxing
mode in which the data of the existing Matroska Blocks is not simply
forwarded as-is. Instead data from several Blocks is recombined
together to output several packets. The parameters governing this
process are parsed from the CodecPrivate: Coded framesize (cfs), frame
size (w) and sub_packet_h (h).

During demuxing, h/2 pieces of data of size cfs each are read from every
Matroska (Simple)Block and put at offset m * 2 * w + n * cfs of a buffer
of size h * w, where m ranges from 0 to h/2 - 1 for each Block while n
is initially zero and incremented after a Block has been parsed until it
is h, at which poin the assembled packets are output and n reset.

The highest offset is given by (h/2 - 1) * 2 * w + (h - 1) * cfs + cfs
while the destination buffer's size is given by h * w. For even h, this
leads to a buffer overflow (and potential segfault) if h * cfs > 2 * w;
for odd h, the condition is h * cfs > 3 * w.

This commit adds a check to rule this out.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

RealAudio 28.8 does not need or use sub_packet_size for its demuxing
and this field is therefore commonly set to zero. But since 18ca491b
the Real Audio specific demuxing is no longer applied if sub_packet_size
is zero because the codepath for cook and ATRAC3 divide by it; this made
these files undecodable.

Furthermore, since 569d18aa (merged in 2c8d876d) sub_packet_size being
zero is used as an indicator for invalid data, so that a file containing
such a track was completely skipped.

This commit fixes this by not checking sub_packet_size for RealAudio
28.8 at all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

They need a special parsing mode and in order to find out whether this
mode is in use, several checks have to be performed. They can all be
combined into one: If the buffer that is only used to assemble their
packets has been allocated, use the RealAudio parsing mode.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

Only flavors 0..3 seem to exist. E.g. rmdec.c treats any flavor > 3
as invalid data. Furthermore, we do not know how big the packets to
create ought to be given that for sipr these values are not read from
the bitstream, but from a table.

Furthermore, flavor is only used for sipr, so only check it for sipr;
rmdec.c does the same. (The old check for flavor being < 0 was
always wrong given that flavor is an int that is read via avio_rb16(),
so it has been removed completely.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Failure was due to the extra comment printed by libavcodec/utils.c since
side data is used to signal the skipped samples.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

Part of this patch is based on Paul B Mahol's patch from last year.

This also allows for single-stream parameter/codec changes.

This cleans up the code and simplifies it.
It also speeds up parsing since the old pb position was incorrect.

This makes decoding far more robust, since OggS, the ogg magic,
can be commonly found randomly in streams, which previously made
the demuxer think there's a new stream or a change in such.

This also makes implementing CRC checking far simpler and more robust.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>

We only care about the right most bit.
Signed-off-by: James Almer <jamrial@gmail.com>

v4l2_m2m uses device memory mapped buffers to store dequeued
frames/packets (reference counted by AVBufferRef). When the reference
count drops to zero, the buffer ownership is returned back to the
device, so that they can re-filled with frames/packets.

There are some cases when all the capture buffers are in userspace
(i.e. due to internal buffering in ffmpeg). On the s5p-mfc this causes
an infinite wait when polling to dequeue the buffers, which can be
prevented by increasing the total number of buffers. This commit adds a
warning when all the capture buffers are dequeued.
Reviewed-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>

Apparently the changes from 3c9185bf
aren't enough; even with that in place, I got errors like this
when trying to build for iOS:

src/libavdevice/avfoundation.m:135:5: error:
      'AVCaptureDeviceTransportControlsPlaybackMode' is unavailable: not
      available on iOS
    AVCaptureDeviceTransportControlsPlaybackMode observed_mode;
    ^

Use A in calculation.