If a PAT is finished while a PMT section filter is opened but
not yet finished, the PMT section filter is closed and all
the received data is discarded.
This is usually not an issue but some multiplexers (With very
quick PAT/PMT repetition settings) consistently emit a PMT
section start, then a PAT, and then the rest of the PMT,
causing the aforementioned behavior to result in no PMT being
finished.
In the most pathologic situation the stream information are lost
and the probe fallback miscategorizes subtitles as mp3 audio.
Avoid the issue through eliminating redundant PSI/SI table
updates by checking their version field, which is required by
the standard to be incremented on every change no matter how
minor.
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes the calculation of the number of needed blocks to make
sure that ALL pixels are represented by the result.
Signed-off-by: Martin Storsjö <martin@martin.st>
Set this field to TRUE if the audio component is to operate on
little-endian data, and FALSE otherwise.
However TRUE and FALSE are not defined. Since this flag is just a boolean,
interpret all values except for 0 as little endian.
Sample-Id: 64bit_FLOAT_Little_Endian.mov
CC: libav-stable@libav.org
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit b8d7f3186e)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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>
Icecast uses HTTP 1.0 while Libav uses HTTP 1.1 and enables by
default chunked post.
Icecast actually forwards the HTTP chunk headers to the listener
as part of the media stream (without the chunk encoding HTTP headers)
causing the players to lose sync.
Disabling the option is enough to feed icecast properly.
(cherry picked from commit 76c70e33d2)
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
In matroska_read_seek(), |tracks| is assigned at the begining of the
function. However, functions like matroska_parse_cues() could reallocate
the tracks and invalidate |tracks|.
This assigns |tracks| only before using it, so that it will not get
invalidated elsewhere.
Bug-Id: chromium/427266
should be the raw amount of pixels (for example 3840x1080 for full HD side by
side) and the DisplayWidth/Height in pixels should be the amount of pixels for
one plane (1920x1080 for that full HD stream)."
So, move the aspect ratio check in the mkv_write_stereo_mode() function
and always write the embl when stereo format and/or aspect ration is set.
Also add a few comments to that function.
CC: libav-stable@libav.org
Found-by: Asan Usipov <asan.usipov@gmail.com>
The new function wraps errno so that its value is correctly reported
when other functions overwrite it (eg. in case of logging).
CC: libav-stable@libav.org
Bug-Id: CID 1135748
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
The quality scale field is only supposed to be present if the fourth bit
is set. In practice, lame always sets it, but other tools might not.
CC:libav-stable@libav.org
Having more than 10 consecutive frames decoded as mp3 should be
considered a clear signal that the sample is mp3 and not mpegps.
Reported-By: Florian Iragne <florian@iragne.fr>
CC: libav-stable@libav.org
If we throw away the buffered incomplete frame, make sure to also
throw away the buffered bits of an incomplete byte at the same
time.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit df07c07b3d)
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
A failure in segment_end() or segment_start() would lead to freeing
a dangling pointer and in general further calls to seg_write_packet()
or to seg_write_trailer() would have the same faulty behaviour.
CC: libav-stable@libav.org
Reported-By: luodalongde@gmail.com
(cherry picked from commit b3f0465736)
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The size variable is (correctly) unsigned, but is passed to several functions
which take signed parameters, such as avio_read, sometimes after having
numbers added to it. So ensure that size remains within the bounds that
these functions can handle.
(cherry picked from commit c5560e72d0)
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Convert the Matroska stereo format to the Stereo3D format, and add a
Stereo3D side data to the stream.
Bump the doctype version supported.
Bug-Id: 728 / https://bugs.debian.org/757185
It is written to the file as a 22-bit value.
CC: libav-stable@libav.org
(cherry picked from commit 75bbaf2493)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
At least one FATE sample contains such chunks and happens to work simply
by accident (due to find_stream_info() swallowing the error).
CC: libav-stable@libav.org
(cherry picked from commit 4d6c515284)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This reverts commit 30e50c5027.
The original commit broke the ability to stream AAC over HTTP/Icecast. It looks
like avformat_find_stream_info() gets stuck in an infinite loop, never hitting
AVFormatContext.max_analyze_duration since duration is never set for any of
the packets.
Example stream: http://listen.classicrocklounge.com:8000/aac64
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Update mxf_set_audio_pts to use the container-provided information.
The UL is marked as "to be changed in the future", but the current
samples in the wild do use it.