aac_adtstoasc makes the aac extradata available only after the first packet
is filtered, and as packet side data.
Assume extradata will be available as part of the first packet if
avpriv_mpeg4audio_get_config() fails the first time due to missing extradata
and reserve space for the OutputSampleRate element in the Tracks master.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This matrix needs to be applied after all others have (currently only
display matrix from trak), but cannot be handled in movie box, since
streams are not allocated yet. So store it in main context, and apply
it when appropriate, that is after parsing the tkhd one.
Fate tests are updated accordingly.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
The use of TLSv1_*_method() disallows newer protocol versions; instead
use SSLv23_*_method() and then explicitly disable the deprecated
protocol versions which should not be supported.
When the macro is expanded with a semicolon following it and the
macro itself contains a semicolon, we ended up in double semicolons,
which is treated as a statement that disallows further declarations.
This avoids errors about mixed declarations and statements on gcc,
after ee05079766.
Signed-off-by: Martin Storsjö <martin@martin.st>
Instead use our own struct, which we already use when using
gcrypt and gnutls.
In OpenSSL 1.1, the DH struct has been made opaque.
Signed-off-by: Martin Storsjö <martin@martin.st>
For 'nclx', the latest edition of the standard switched from JPEG XR
to 23001-8, which matches the current order of our entries. Bounds
are preserved as a sanity check.
For 'nclc', qtff edition 2016-09-13 introduced a few new entries.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This also fixes a minor bug introduced in the codecpar conversion, where
the termination condition for extracting the extradata does not match
the actual extradata setting code. As a result, the packet durations
made up by lavf go back to their values before the codecpar conversion.
That is of little consequence since that code should eventually be
dropped completely.
This way they can be reused by other code without including the whole
decoder-specific hevcdec.h
Also, add the HEVC_ prefix to them, since similarly named values exist
for H.264 as well and are sometimes used in the same code.
The spec says
9: Interlaced with bottom field displayed first and top field stored first
14: Interlaced with top field displayed first and bottom field stored first
And avcodec.h states
AV_FIELD_TB, //< Top coded first, bottom displayed first
AV_FIELD_BT, //< Bottom coded first, top displayed first
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
According to the public RTMP specification, these 4 bytes should
be zero.
librtmp in server mode assumes that the RTMPE (FP9) handshake is
used if these bytes are nonzero.
Signed-off-by: Martin Storsjö <martin@martin.st>
When acting as server, the server can include a "clientid" property
in some status messages. But this should be a unique number
identifying the client session, not identifying the server itself.
In practice, omitting it works just as well as including this
incorrect field.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that e.g. Adobe FME actually reacts to it. As long
as the value we've been sending is the default one (128), the bug
hasn't been noticed.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some applications such as Adobe FME append lots of parameters
here, making it easily overflow the current limit.
Signed-off-by: Martin Storsjö <martin@martin.st>
It is supposed to be a flag. The only currently defined value is
AVIO_SEEKABLE_NORMAL, but other ones may be added in the future.
However all the current lavf code treats this field as a bool (mainly
for historical reasons).
Change all those cases to properly check for AVIO_SEEKABLE_NORMAL.
This was introduced in bc2a32969e.
The whole block that the statement was added to is only
relevant when used as a demuxer, but the other statements
there have had other if statements guarding them. Make
sure to only run this whole block if being used as a
demuxer.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This was added before edts support existed, and is no longer
valid.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This breaks files with legitimate single-entry edit lists,
and the hack, introduced in f03a081df0,
has no link to any known sample in its commit message.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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.
