Previous "currently size is limited" didn't give away
much in terms of useful info.
Signed-off-by: Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
So far an AC-3 elementary stream is refered to in the PMT according to
System A (ATSC). An E-AC-3 ES in contrast is embedded the System B (DVB) way.
To fix this inconsistency, this commit changes the default E-AC-3 behaviour to
use the ATSC way, too. Furthermore a new flag is added to optionally select the
DVB way (regarding both codecs and possible further differences in the future).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes a specific srt sample, which has an event with negative duration.
libavcodec will convert an event with negative duration to an ASS event
which will be displayed forever, which is not wanted here.
Treat negative duration always as unknown duration instead, and show it
until the next subtitle event.
A negative bits_per_coded_sample doesn't make sense.
If it is too large, the size calculation for av_get_packet overflows,
resulting in allocation of a too small buffer.
Also make sure width and height are sane.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes NULL pointer dereferencing.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
If it is negative, it can cause the byte position to move backwards in
avio_skip, which in turn makes sm_size negative and thus size larger
than the size of the packet buffer, causing invalid writes in avio_read.
Also fix potential overflow of avio_tell(bc) + value_len.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Fixes ubsan runtime error: null pointer passed as argument 2, which is
declared to never be null
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Since the parser was merged back almost 2 months ago this is the first
time the bitstream of the container has been updated.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* commit 'e02de9df4b218bd6e1e927b67fd4075741545688':
lavc: export Dirac parsing API used by the ogg demuxer as public
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
A negative codec_id cannot be handled by the found_decoder API of
AVStream->info: if the codec_id is not recognized, found_decoder is set
to -codec_id, which has to be '<0' according to the API documentation.
This can cause NULL pointer dereferencing in try_decode_frame.
Also make sure the codec_type matches the expected one for codec_id.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
support writing encrypted mp4 using aes-ctr, conforming to ISO/IEC
23001-7.
3 new parameters were added:
- encryption_scheme - allowed values are none (default) and cenc-aes-ctr
- encryption_key - 128 bit encryption key (hex)
- encryption_kid - 128 bit encryption key identifier (hex)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
So far an AC-3 elementary stream is refered to in the PMT according to
System A (ATSC). However System B (DVB) has a different way to signal an AC-3
ES within the PMT. This different way can be enabled by a new flag. The flag is
more generally named 'system_b' as there are further differences between ATSC
and DVB (e.g. the signalling of E-AC-3) which should then also be covered by it
in the future.
Bug-Id: 73
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Similar to 33fefdb44.
Fix trac ticket #4921.
Signed-off-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The current muxer behaviour is to create streams in read_header() based
on the audio/video presence flags, but fill in the stream parameters
later when we actually get some packets for them. This is rather shady,
since other demuxers set the stream parameters immediately when the
stream is created and do not touch the stream codec context after that.
Change the flv demuxer to behave in the same way as other similar
demuxers -- create the streams only when we get a packet for them.
Almost all the places from which this function is called already check
the header manually and in the two that don't (the mp3 muxer) the check
should not cause any problems.
It will not be set unless the muxing codec context is also the encoding
context, which is discouraged. When the frame size is not known from
av_get_audio_frame_duration(), the fallback should still be good enough.
It will not be set if the stream codec context is not the encoding
context. Use av_get_audio_frame_duration() instead, it should work for
all audio codecs supported by the muxer.
