It is used in adx_read_packet, which currently depends on the
decoder/parser setting this value between reading the file header and
demuxing the first packet.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The chunk size is limited to UINT16_MAX (written by avio_wb16), so make
sure that the packet size is not too large.
Such large frames need to be split into slices smaller than 64 kB, but
that is currently supported neither by the rv10/rv20 encoders nor the rm
muxer.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
According to the WebP Lossless Bitstream Specification the highest
allowed value for a prefix code is 39.
If prefix_code is too large, the calculated extra_bits has an invalid
value and triggers an assertion in get_bits.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If it doesn't fit into 12 bits it triggers an assertion.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The original flags variable contains rtpdec flags, while the
rmflags variable contains RM flag bits which have a completely
different definition.
Signed-off-by: Martin Storsjö <martin@martin.st>
The only case where RTP_FLAG_KEY actually is needed is
in RDT, where such a flag needs to be passed via the
rtpdec parse function's flags parameter.
Signed-off-by: Martin Storsjö <martin@martin.st>
Nothing in the framework nor in the rest of the depacketizer actually
uses this flag - the chained demuxer sets the keyframe flag properly on
demuxed packets already.
Signed-off-by: Martin Storsjö <martin@martin.st>
Although the specification mandates this bit to zero, it may happen
that software tools incorrectly flip it to one, invalidating a possibly
valid stream.
Relax this restriction, by failing only when AV_EF_BITSTREAM is set.
This behaviour is similar to aac decoders in Firefox and Quicktime.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This fixes an oversight in 96084251, in a refactoring done on top
of Gilles' original patch.
Pointed out by Gilles Chanteperdrix.
Signed-off-by: Martin Storsjö <martin@martin.st>
Similarly to what has been done for MOV, display XMP metadata only when
users explicitly require it.
The Extensible Metadata Platform tag can contain various kind of data
which are not strictly related to the video file, such as history of
edits and saves from the project file.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Normally the aic decoder finds the proper slice combination (multiple of
some number less than 32) but in case of odd width, it resorts to the
default values, which were actually swapped.
The number of slices is modified to account for such odd width cases.
CC: libav-stable@libav.org
This reverts commit 4abfa387b8.
This commit broke playback of fragmented mp4 files with b-frames.
While investigating this, it turned out that the general framework
isn't ready for a PTS-based index yet. Revert this change until
a better thought out solution is in place.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some files produced by the official encoder have up to 16bit of
padding instead of the expected padding to the byte.
Use a self-explanatory macro instead of a simple number.
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Instead check the timestamps while muxing, to avoid buffering a
too long timestamp range into one single packet.
This makes the AMR and AAC packetization slightly less efficient,
since we set a possibly unnecessarily high max_frames_per_packet.
(These packetizers end up doing a memmove of the TOC bytes if
sending a packet before max_frames_per_packet is achieved, and
we end up setting max_frames_per_packet to a value that should
be high enough for most uses.)
All packetizers that use max_frames_per_packet now set it either
to a default value, or to a value calculated based on other
parameters, so none of them rely on the previous default setting.
For iLBC, copy one frame at a time, to allow checking the timestamp
range for each of them - basically doing potentially multiple
loops to simplify the code instead of trying to calculate the
number of frames to buffer while honoring s1->max_delay.
This is in preparation for reducing the coupling between libavformat
and libavcodec, by not having the muxers use the encoder field
frame_size (which may not be available during e.g. stream copy).
Signed-off-by: Martin Storsjö <martin@martin.st>
Factorize out the s->num_frames check at the start of the if statements,
simplifying adding more alternative causes for sending the buffered
frames.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids having to jump to the defaultcase in the switch. Manually
override the stream time base back to 90 kHz for the few audio codecs
that don't use the sample rate as time base (mp2, mp3).
Signed-off-by: Martin Storsjö <martin@martin.st>
This doesn't fix any bug, but makes the code simpler for later
patches, and more straightforward to read as is.
Signed-off-by: Martin Storsjö <martin@martin.st>
After sending a fragmented frame, len (s->buf_ptr - s->buf) isn't
zero, while s->num_frames is zero as intended. Using s->num_frames
makes it work as intended, and is less convoluted than keeping track
of (resetting) s->buf_ptr.
This avoids sending stray data after sending a fragmented aac packet.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Don't prefix them ffio_url, which is misleading, sounding too
much like the urlprotocol layer (like ffurl_*).
Signed-off-by: Martin Storsjö <martin@martin.st>
Incidentally `-y` also collides with avconv global options.
Update x11grab to match and document the option.
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Anton Khirnov
Andreas Cadhalpun
Thomas Volkert
Himangi Saraogi
Martin Storsjö
Luca Barbato
Vittorio Giovara
Diego Biurrun
Christian Hujer
John Stebbins