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.
This seems not to do anything any more since a long time, and removing
it avoids using uninitialized memory. Also change the error value
forwarding as done everywhere else.
Partly fixes: msan_uninit-mem_7fb7d24780d0_2744_R03T.CAK
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
I've got some m4a samples that had jpeg cover art marked as png. Since
these files were supposedly written by iTunes, and other software can
read it (e.g. clementine does), this should be worked around.
Since png has a very simple to detect header, while it's apparently a
real pain to detect jpeg in the general case, try to detect png and
assume jpeg otherwise. Not bothering with bmp, as I have no test case.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Some codecs use the codec_tag to signal specific information and
picking the first one would lead to a broken file.
Bug-Id: 883
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Some systems may be lacking getservbyport; the previous ifdef wasn't
quite enough since it still assumed that struct servent was defined,
as pointed out by Clément Gregoire.
Simply remove the possibility to return non-numeric services in
getnameinfo; no caller of getnameinfo within libavformat
currently try to use getnameinfo for retrieving the port number without
NI_NUMERICSERV, and falling back on getservbyport may be non-threadsafe.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some server in the wild do not put the boundary at a newline
as rfc1347 7.2.1 states.
Cope with that by reading a line and if it is not empty reading
a second one.
Reported-By: bitingsock
This also makes sure that a fragmented file without the empty_moov
flag (i.e. with a non-empty initial moov fragment) actually gets
written, if some of the tracks turn out to not have any samples.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some RTSP servers ("HiIpcam/V100R003 VodServer/1.0.0") respond to
our keepalive GET_PARAMETER request by a truncated RTSP header
(lacking the final empty line to indicate a complete response
header). Prior to 764ec70149, this worked just fine since we
reacted to the $ as interleaved packet indicator anywhere.
Since $ is a valid character within the response header lines,
764ec70149 changed it to be ignored there. But to keep
compatibility with such broken servers, we need to at least
allow reacting to it at the start of lines.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes access to Grandstream cameras, which return 401 otherwise.
VLC sends Authorization: header with spaces between parameters, and it
is known to work with Grandstream devices and broad range of other HTTP
and RTSP servers, so author considers switching to such behaviour safe.
See RFC 2617 (HTTP Auth).
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
In one case it was written as zero, one case left it uninitialized,
missed the 11 bytes for the flv header.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
"language" is not an offical matroska tag.
Track languages are specified with the MATROSKA_ID_TRACKLANGUAGE ebml.
Writing the tag overrides the ebml specified language during playback with
libav and some other players.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Note that convergence_duration had another meaning, one which was in
practice never used. The only real use for it was a 64 bit replacement
for the duration field. It's better just to make duration 64 bits, and
to get rid of it.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Note that this slightly changes behavior: it sets AVMEDIA_TYPE_UNKNOWN
if the codec type is unknown. This should be ok.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
packets are queued due to packet reordering until the queue reach its
maximal size or max delay is reached.
This commit adds a warning trace when max delay is reached.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit print as AV_LOG_VERBOSE the jitter buffer
size. It might be the default value or the value set by application.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit adds a warning trace when jitter buffer
is full. It helps to understand leading decoding issues.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Since the actual max length of the jitter buffer is restricted by
max_delay, this shouldn't harm the overall latency (assuming that
max_delay is set properly), while allowing packet reordering with
a larger number of packets (which may be required with high bitrate
video).
Signed-off-by: Martin Storsjö <martin@martin.st>
The previous restriction was partially designed to fix certain
(broken) samples from bug 215. There should be no restriction on the
number of keyframes per fragment or trun.
The spec suggests that all frames lacking MOV_FRAG_SAMPLE_FLAG_IS_NON_SYNC
are key frames, but we require the flag MOV_FRAG_SAMPLE_FLAG_DEPENDS_YES
to be unset as well. This works for (possibly broken) media that never
sets the NON_SYNC flag and should also be correct for any spec-compliant
file.
For files that never set either of the flags, all samples are marked
as keyframes.
Signed-off-by: Martin Storsjö <martin@martin.st>
And update the preference for the newer codecs now that the libraries
seem stable and widespread enough.
Bug-Id: 695
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
also do not return the error code but just break reading
metadata object in the case of the aspect ratio reading failure
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The double meaning of the faststart flag (moving a moov atom
to the start of files, making them streamable, for non-fragmented
files, vs inserting a global sidx index at the start of files
for fragmented files) is confusing - see 40ed1cbf1 for
explanation of its origins.
Since the second meaning of the flag hasn't been part of any
libav release yet, just rename it to get rid of the confusion
without any extra deprecation (which wouldn't get rid of the
potential confusion, of users adding -movflags faststart
even for fragmented files, where it isn't needed for making
them "streamable").
This gets back the old behaviour, where -movflags faststart
doesn't have any effect for fragmented files.
Signed-off-by: Martin Storsjö <martin@martin.st>
For fragmented files with non-empty moov, with a fragment index
(sidx), place the index after the initial moov/mdat pair.
Previously, for this pathological case, the index was written
at the start of the file.
Signed-off-by: Martin Storsjö <martin@martin.st>
The same field is also used for writing the sidx index header,
for fragmented files, when the faststart flag is used.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes crashes with pathological cases when trying to write
a sidx index (via the -movflags faststart option, in combination
with fragmenting options), when no fragments actually have been
written. (This is possible if the empty_moov flag isn't used,
so that all actual packet data is written in the moov/mdat pair,
and no moof/mdat pairs have been written.)
In these pathological cases, no sidx should be written at all.
Signed-off-by: Martin Storsjö <martin@martin.st>
The length of BOOL values is 16 bits in the Metadata Object but
32 bits in the Extended Content Description Object.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Allow $ as character anywhere within normal RTSP replies - both
within the lines, and as the first character of RTSP header lines.
(The existing old comment indicated that an inline packet could
start at any line within a RTSP reply header, but that doesn't
sound valid to me, and I'm not sure if the existing code
handled that correctly either.)
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
this condition breaks reading from the pipe as data_reached variable
have to be set to break while in the asf_read_header just after the Data
Object is found
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This is necessary to preserve the quality information currently exported
with coded_frame. Add the new side data to every encoder that needs it,
and use it in avconv.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Fix warning from clang "absolute value function 'abs' given an argument
of type 'long long' but has parameter of type 'int' which may cause
truncation of value [-Wabsolute-value]".
The old one is the result of the reverse engineering and guesswork.
The new one has been written following the now-available specification.
This work is part of Outreach Program for Women Summer 2014 activities
for the Libav project.
The fate references had to be changed because the old demuxer truncates
the last frame in some cases, the new one handles it properly.
The seek-test reference is changed because seeking works differently
in the new demuxer. When seeking, the packet is not read from the stream
directly, but it is rather constructed by the demuxer. That is why
position is -1 now in the reference.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
14496-3 suggests packing main_data of MP3 that is usually scattered
into multiple frames due to bit reservoir.
However, after packing main_data into a access unit, bitrate index
in the MPEG audio frame header doesn't match with actual frame size.
In order to accept this, this patch removes unnecessary frame size
checking on mp3 decoder.
Also, mov demuxer was changed to use MP3 parser only on special cases
(QT MOV with specific sample description) to avoid re-packetizing.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
GNUTLS_SHUT_RDWR means GnuTLS will keep waiting for the server's
termination reply. But since we don't shutdown the TCP connection at
this point yet, GnuTLS will just keep skipping actual data from the
server, which basically is perceived as hang.
Use GNUTLS_SHUT_WR instead, which doesn't have this problem.
Signed-off-by: Martin Storsjö <martin@martin.st>
display_matrix_size is only initialized when av_stream_get_side_data()
returns a side data pointer. The code is safe since the only effect this
has is setting the display_matrix pointer to NULL which it was already
anyway.
The first check is done without the AVIOContext, so alloc it only if said check succeeds
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
mpz_import and mpz_export were added in GMP 4.1, in 2002.
This simplifies the DH code by clarifying that it only uses pure
bignum functions, no other parts of nettle/hogweed.
Signed-off-by: Martin Storsjö <martin@martin.st>
If avio_read fails, the buffer can contain uninitialized data.
This fixes 'Conditional jump or move depends on uninitialised value(s)'
valgrind warnings, and addresses a few memleaks.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
In this case the mov demuxer can return a large number of empty packets.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Otherwise the loop can take a lot of time if num_descr is very large.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
OSX does not know MSG_NOSIGNAL. BSD (which OSX is based on) has got
the socket option SO_NOSIGPIPE (even if modern BSDs also support
MSG_NOSIGNAL).
Signed-off-by: Martin Storsjö <martin@martin.st>
There was a misunderstanding betewen bits and bytes for the parameter
value for generating random big numbers.
Signed-off-by: Martin Storsjö <martin@martin.st>
Move the OpenSSL and GnuTLS implementations to their own files. Other
than the connection code (including options) and some boilerplate, no
code is actually shared.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since the underlying URLContext read functions are used,
they handle interruption, without having to handle it at
this level.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids hijacking the fd, by reading using the normal
URLContext functions instead. This allowing reading data that has
been buffered in the underlying URLContext.
This avoids using the libraries own send functions that can
cause SIGPIPE.
The fd is still used for polling the lowlevel socket, for
waiting for retries.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that the time + duration of the first segment
matches the start time of the next segment for e.g. AAC audio
with encoder delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
