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>
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.
matroskaenc applies divisors to the display width/height when generating
stereo content. This patch adds the corresponding multipliers to matroskadec
so that the original sample aspect ratio can be recovered.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Checking the codec context parameters to find out this information is
far too unreliable to be useful, so it is safer to assume B-frames are
always present.
The demuxer returned INVALIDDATA and failed to demux the remaining data
when an invalid stream index was read, now it just skips the asf packet
for the stream with an invalid stream index and continues demuxing.
Reported-By: Hendrik Leppkes
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This feature allows making associations between audio tracks
that apple players recognize. E.g. when an ac3 track has a
tref that points to an aac track, devices that don't support
ac3 will automatically fall back to the aac track.
Apple used to *guess* these associations, but new products
(AppleTV 4) no longer guess and this association can only
be made explicitly now using the "fall" tref.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The XTEA algorithm operates on 32 bit numbers, not on byte sequences.
The XTEA implementation in libavutil is written assuming big endian
numbers, while the rtmpe signature encryption assumes little endian.
This fixes rtmpe communication with rtmpe servers that use signature
type 8 (XTEA), e.g. crunchyroll.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some entries might be either empty or contain types we do not parse
(eg. 'url '). In both cases, if an 'alis' is not the first entry,
external references are not loaded, so make sure that the array starts
with an 'alis' dref.
Rather than reading the alternate absolute path version from dref
type 18, make sure that 0s are considered as '/'. These values are
sometimes present in the full path, and are mistakenly interpreted as
line terminators othewise.
With the correct handling of this dref type, parsing type 18 is not
needed any more.
By writing a zero-sized packet, the caller can communicate the
start_dts/start_cts for the stream without actually writing
the first packet.
This allows doing random-access writing of fragments when the
start dts of the stream isn't zero, so that the edit list in the moov
is written based on timestamps from the nominal start time signaled
via the zero-sized packet, while the first proper packet written
corresponds to a later fragment.
To avoid potential unexpected behaviour, empty packets only set
start_dts if the frag_discont flag is set.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows producing fragments discontinously where the video
stream has b-frames (but starts at pts=0), but doesn't work for the
cases with audio with preroll.
Signed-off-by: Martin Storsjö <martin@martin.st>
Contrary to the normal fate tests that run via avconv, this tests
nontrivial call sequences that are only doable via the API
(mainly for different corner cases when using the muxer for
segmenting).
The test muxes fake packet data (with extradata that looks
enough like proper data to make the file be viewable with e.g.
boxdumper) and checks the hash of the produced files. The test also
verifies that fragments produced via different call sequences remain
identical (to avoid e.g. updating the output hashes and suddenly
having fragments that used to be identical suddenly diverging), for
fragments written with frag_discont and/or delay_moov.
Signed-off-by: Martin Storsjö <martin@martin.st>
In most other cases when writing fragmented mp4 files, the output
IO context is flushed after each fragment. Also flush it after
writing the initial moov, to have it behave in the same way.
Signed-off-by: Martin Storsjö <martin@martin.st>
All encoders set pts and dts properly now (and have been doing that for
a while), so there is no good reason to do any timestamp guessing in the
muxer.
The newly added AVStreamInternal will be later used for storing all the
private fields currently living in AVStream.
