This reverts commit 25bacd0a0c.
Since 230b1c070, the bytewise AV_W*() macros only expand their
argument once, so revert to the more readable version of these.
Signed-off-by: Martin Storsjö <martin@martin.st>
AV_WB32 can be implemented as a macro that expands its parameters
multiple times (in case AV_HAVE_FAST_UNALIGNED isn't set and the
compiler doesn't support GCC attributes); make sure not to read
multiple times from the source in this case.
Signed-off-by: Martin Storsjö <martin@martin.st>
There are samples with invalid stsc that may work fine as is and
do not need extradata change. So ignore any out of range index, and
error out only when explode is set.
Found-by: Matthieu Bouron <matthieu.bouron@stupeflix.com>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
When writing a fragmented file, we by default write an index pointing
to all the fragments at the end of the file. This causes constantly
increasing memory usage during the muxing. For live streams, the
index might not be useful at all.
A similar fragment index is written (but at the start of the file) if
the global_sidx flag is set. If ism_lookahead is set, we need to keep
data about the last ism_lookahead+1 fragments.
If no fragment index is to be written, we don't need to store information
about all fragments, avoiding increasing the memory consumption
linearly with the muxing runtime.
This fixes out of memory situations with long live mp4 streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
This function needs to return false, or data in the additional tables
will be skipped, and the decoder will not be able to decode frames
associated with them.
Store data from each stsd in a separate extradata buffer, keep track of
the stsc index for read and seek operations, switch buffers when the
index differs. Decoder is notified with an AV_PKT_DATA_NEW_EXTRADATA
packet side data.
Since H264 supports this notification, and can be reset midstream, enable
this feature only for multiple avcC's. All other stsd types (such as
hvc1 and hev1) need decoder-side changes, so they are left disabled for
now.
This is implemented only in non-fragmented MOVs.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This avoids the danger that get_bits.h might get indirectly #included before
BITSTREAM_READER_LE is defined.
Also sort headers into canonical order where appropriate.
Split version files into one line per symbol/directive to allow compatibility
with the Solaris linker without preprocessing and eliminate $ from version file
templates to simplify the postprocessing shell command.
Previously, we required the minimum number of bytes required for
the full box. Don't strictly require the astronomical body and additional
notes fields, but do require an altitude field (which currently isn't
parsed). This matches the initial length check at the start of the function
(which doesn't know about the variable length place field).
Signed-off-by: Martin Storsjö <martin@martin.st>
This was missed in e1eb0fc960, when ff_interleaved_peek was
changed to include const during the evolution of the patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
As long as caller only writes packets using av_interleaved_write_frame
with no manual flushing, this should allow us to always have accurate
durations at the end of fragments, since there should be at least
one queued packet in each stream (except for the stream where the
current packet is being written, but if the muxer itself does the
cutting of fragments, it also has info about the next packet for that
stream).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows callers with avio write callbacks to get the bytestream
positions that correspond to keyframes, suitable for live streaming.
In the simplest form, a caller could expect that a header is written
to the bytestream during the avformat_write_header, and the data
output to the avio context during e.g. av_write_frame corresponds
exactly to the current packet passed in.
When combined with av_interleaved_write_frame, and with muxers that
do buffering (most muxers that do some sort of fragmenting or
clustering), the mapping from input data to bytestream positions
is nontrivial.
This allows callers to get directly information about what part
of the bytestream is what, without having to resort to assumptions
about the muxer behaviour.
One keyframe/fragment/block can still be split into multiple (if
they are larger than the aviocontext buffer), which would call
the callback with e.g. AVIO_DATA_MARKER_SYNC_POINT, followed by
AVIO_DATA_MARKER_UNKNOWN for the second time it is called with
the following data.
Signed-off-by: Martin Storsjö <martin@martin.st>
When feeding input RTP packets to the depacketizer via custom IO,
it needs to pick the right stream using the payload type for
RTP packets, and using the SSRC for RTCP packets. If the first
packet is an RTCP packet, we don't (currently) know the SSRC
yet and thus can't pick the right RTP depacketizer to handle it.
By parsing the SSRC attribute in the SDP, we can map initial
RTCP packets to the right stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
It doesn't matter what the actual reason for not returning
an AVPacket was - if we didn't return any packet and we have
the next one queued, parse it immediately. (rtp_parse_queued_packet
always consumes a queued packet if one exists, so there's no risk
for infinite loops.)
Signed-off-by: Martin Storsjö <martin@martin.st>
The declarations that this comment referred to were removed
in 2439f2ca8 - there is no unbuffered IO in this header now.
Signed-off-by: Martin Storsjö <martin@martin.st>
We still only support one single layer though, but this allows
receiving streams that have this structure present even for
single layer streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
This codepath isn't quite as bad as it used to sound, if fragments
are cut automatically at video packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Restore alphabetical order in lists, break overly long lines, do some
prettyprinting, add some explanatory section comments, group parts
together that belong together logically.
The problem is that the argument 'q' is of the type uint8_t.
According to the JPEG standard, if 1 <= q <= 50, the scale factor
'S' should be 5000 / Q. Because the create_default_qtables() reuses
the variable 'q' to store the result of this calculation, for small
values of q < 19, q wil subsequently overflow and give wrong results
in the calculated quantization tables.
Instead, use a new variable 'S' (same name as in RFC2435) with the
proper range to store the result of the division.
Signed-off-by: Martin Storsjö <martin@martin.st>
Apply the default value for timeout in code instead of via the
avoption, to allow distinguishing the default value from the user
not setting anything at all.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.
This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).
Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.
Signed-off-by: Martin Storsjö <martin@martin.st>
Using this requires setting the rw_timeout option to make it
terminate, alternatively using the interrupt callback (if used via
the API).
Signed-off-by: Martin Storsjö <martin@martin.st>
If set non-zero, this limits duration of the retry_transfer_wrapper()
loop, thus affecting ffurl_read*(), ffurl_write(). As soon as
one single byte is successfully received/transmitted, the timer
restarts.
This has further changes by Michael Niedermayer and Martin Storsjö.
Signed-off-by: Martin Storsjö <martin@martin.st>
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Check if the size is written the first 4 bytes and read the next 4
as fourcc candidate, fallback checking the initial for 4 bytes.
"The CodecPrivate contains all additional data that is stored in the
'stsd' (sample description) atom in the QuickTime file after the
mandatory video descriptor structure (starting with the size and FourCC
fields)"
CC: libav-stable@libav.org
Samples produced by Omneon (Harmonic) store external references with
paths ending with 0s. Such movs cannot be loaded properly since every
0 is converted to '/', to keep the same parsing code for dref type 2
and type 18: this makes the external reference point to a non-existing
direactory, rather than to the actual referenced file.
Add a brief trimming loop that drops all ending 0s before trying to
parse the external reference path.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Store the file duration in the same timebase it arrives (i.e.
milliseconds) and only convert it to the file duration units (100ns)
when it's actually written, thus simplifying some calculations. Also,
store the duration as unsigned, since it cannot be negative.
CC: libav-stable@libav.org
Bug-ID: CVE-2016-2326
Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
Instead of a linked list constructed at av_register_all(), store them
in a constant array of pointers.
Since no registration is necessary now, this removes some global state
from lavf. This will also allow the urlprotocol layer caller to limit
the available protocols in a simple and flexible way in the following
commits.
Some muxer might or might not fit incomplete mp3 frames in
their packets.
Bug-Id: 899
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes infinite loops due to seeking back.
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes infinite loops due to seeking back.
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The loop can be very long, even though the file is very short.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
asf_read_payload can unset eof_reached, so check it also before calling
that function.
This fixes infinite loops.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Some (de)muxers open additional files beyond the main IO context.
Currently, they call avio_open() directly, which prevents the caller
from using custom IO for such streams.
This commit adds callbacks to AVFormatContext that default to
avio_open2()/avio_close(), but can be overridden by the caller. All
muxers and demuxers using AVIO are switched to using those callbacks
instead of calling avio_open()/avio_close() directly.
(de)muxers that use the URLProtocol layer directly instead of AVIO
remain unconverted for now. This should be fixed in later commits.
This feature is mostly only used by NLE software, and is
both of dubious value being enabled by default, and a
possible security risk.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
For http, this avoids spurious warnings about failed requests (e.g.
HTTP error 416 Requested Range Not Satisfiable), if the last packet
is truncated and the size read is bogus.
Signed-off-by: Martin Storsjö <martin@martin.st>
When loading a truncated flv file, it would previously try to do a seek to
the end of every packet read. For some input protocols (such as http), such
repeated seek attempts are cripple the reading performance.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes runtime error: null pointer passed as argument 2, which is
declared to never be null
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Both avio_skip and detect_unknown_subobject use int64_t for the size
parameter.
This fixes a segmentation fault due to infinite recursion.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Otherwise invalid values are used unchecked in the next run.
This can cause NULL pointer dereferencing.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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.
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>
