If an URLContext is passed in, its ownership is given to this
function, and is either owned by the returned AVFormatContext
on a successful return, or freed on failure.
Signed-off-by: Martin Storsjö <martin@martin.st>
The sample_rate variable is used for checks for audio format
changes at the end of the function.
This fixes cases where the sample rate was set from the codec
id by flv_set_audio_codec (as for nellymoser 8 kHz/16 kHz),
so the value set to last_sample_rate wasn't equal to sample_rate
at this point. This caused the demuxer otherwise reports a spurious
change to 5512 Hz and back to the correct one.
Updating channels in the same way is only done for consistency.
Currently, flv_set_audio_codec doesn't update that value.
Signed-off-by: Martin Storsjö <martin@martin.st>
tcp_shutdown() isn't needed at the moment, but is added for
consistency to explain how the function is supposed to be used.
Signed-off-by: Martin Storsjö <martin@martin.st>
If using the new -rtmp_app and -rtmp_playpath parameters,
one can in many cases set the main url to just rtmp://server/.
If the trailing slash is omitted, path is a string of zero length,
and using path+1 will end up reading uninitialized data.
Signed-off-by: Martin Storsjö <martin@martin.st>
According to the behaviour of librtmp, it is recommended to send this
message to the server after receiving the 'onBWDone' callback in order
to do bandwidth checking and improve compatibility with some servers.
The strtol() interface makes it difficult to use with
const-qualified pointers. With this change, although
the const is still lost, the compiler does not warn
about it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add/fix spacing, split long lines, align assignments where suitable.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Split long comments, move long comments at the end of lines to
separate lines above, fix vertical alignment, fix up comment style
(unify trailing dots - comments had a mix of 2, 3 or 4 dots, where
it would be just as good without them at all).
Signed-off-by: Martin Storsjö <martin@martin.st>
It is worth keeping instead of removing, in case reading this
bit becomes necessary at some later point.
Signed-off-by: Martin Storsjö <martin@martin.st>
Skip to parse fields for additional independent substreams and its
associated dependent substreams since libavcodec's E-AC-3 decoder does not
support them yet.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
This fixes crashes, where the demuxer could return 0 even
if the returned AVPacket isn't initialized at all. This
could happen if running into EOF or running out of probesize
with non-seekable sources.
Signed-off-by: Martin Storsjö <martin@martin.st>
The new incremental parser doesn't always clear prev_pkt,
however the packet queue is cleared when seeking. Which leads
to a use-after-free.
Verified using Valgrind.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Reduces the amount of upfront data required for cluster parsing
thus decreasing latency on seek and startup.
The change in the seek-lavf_mkv FATE test is due to incremental
parsing no longer reading as much data as the old parser and
thus not having that additional data to generate index entries
based on keyframes. Index entries are added correctly as the
file is parsed.
All FATE tests pass and Chrome has been using this patch for ~6
months without issue.
Currently incremental parsing is not supported for files with
SSA tracks since they require merging packets between clusters.
In this case the code falls back to non-incremental parsing.
Signed-off-by: Aaron Colwell <acolwell@chromium.org>
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The mpegts demuxer reads 5 KB at startup just for discovering
the packet size. Since the default avio buffer size is 32 KB,
the seek back to the start will in most cases be within the
avio buffer, and will in most cases succeed even if the actual
protocol isn't seekable.
This makes the demuxer startup faster/with less data when
reading data from a non-seekable input, by not skipping
the first few KB.
If it fails, don't warn if the protocol isn't seekable, making
it behave as before in the failure case.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fix this warning:
libavformat/aviobuf.c:663:20: warning: assignment discards qualifiers from pointer target type
Although this is a public header, it should remain source and
binary compatible.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If a video track specifies a zero frame rate (invalid but occurs),
this results in a division by zero and subsequent undefined conversion
to integer. Setting the default duration from the frame rate only
if the latter is greater than zero avoids such problems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This option is the stream identifier to play or to publish.
Sometimes the URL parser cannot determine the correct
playpath automatically, so it must be given explicitly
using this option (ie. -rtmp_playpath).
Signed-off-by: Martin Storsjö <martin@martin.st>
This option is the name of application to connect on the RTMP server.
Sometimes the URL parser cannot determine the app name automatically,
so it must be given explicitly using this option (ie. -rtmp_app).
Signed-off-by: Martin Storsjö <martin@martin.st>
Instead of allocating over the original, free first. MOVStreamContext
is zero initialized so no double free will occur. Same style as other
fixes for the same problem in this file.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
During error conditions matroska_parse_block may exit without
freeing the memory allocated for laces.
Found via valgrind: http://pastebin.com/E54k8QFU
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Share the formerly internal write_packet with the hinter and move the
fragment flush logic to the user facing one since it is not concerned
about movtrack-only streams.
Fixes bug #263
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Searching for packet markers doesn't make sense for this use case,
where packets are fed one at a time to the demuxer.
This fixes playing back streams that have packets not starting
with the 0x82, 0x00, 0x00 marker.
Signed-off-by: Martin Storsjö <martin@martin.st>
It can take a long time before subtitles or data streams show up,
so we shouldn't wait for those before assuming we have all info
for streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
The audio codecs property is composed by all values except
SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010) which are
unused.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
When set, if an Ogg stream buffer has enough data, a page is made
instead of filling maximum-size pages. Using smaller pages results
smaller seek intervals at the expense of higher container overhead.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
This patch allows the user to force flushing of all queued packets
by calling av_interleaved_write_frame() with pkt set to NULL.
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Martin Storsjö <martin@martin.st>
This isn't exactly equivalent with the earlier code for codecs
other than H264 and VC1, but those are two only codecs supported
by this codepath anyway, and it simplifies it a bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.
The other fragmentation options (frag_duration, frag_size and
frag_keyframe) are combined with OR, cutting fragments at the
first of the conditions being fulfilled.
Signed-off-by: Martin Storsjö <martin@martin.st>
Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
The private option has not been part of any release yet (and
it is only of use in quite rare cases), so just remove it instead
of keeping it with deprecation warnings.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was forgotten in the transition from av_open_input_file to
avformat_open_input, see 603b8bc2a1.
This doesn't change anything for the default case where the
option isn't set, since PROBE_BUF_MAX is 1048576 (which was
used as max probe size earlier) while the default value for
the probesize option is 5000000, which for the probe function
is clipped to PROBE_BUF_MAX anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
Allow up to 4 retries for normal requests, where both the
proxy and the target server might need to authenticate.
Signed-off-by: Martin Storsjö <martin@martin.st>
These commands are sent asynchronously, not waiting for the reply.
This reply is parsed later by ff_rtsp_tcp_read_packet or
udp_read_packet. If the reply indicates that we used stale
authentication and need to use a new nonce, resend a new keepalive
command immediately.
This is the only request sent asynchronously, so currently there's
no other command that needs to be resent in the same way.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes sending back RTCP RR packets if receiving RTP over
multicast.
If the multicast stream is sent on demand (set up and signalled
via RTSP), the sender might depend on getting RTCP RR packets
knowing that there are listeners, otherwise the stream can be
closed after a certain timeout.
This fixes receiving RTSP streams over multicast on unix, from
certain Axis cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this code was added in 36b532815c, the new code was added
between the existing comment and the existing line of code, making
the old comment seem to refer to the new code. This makes it read
correctly.
Signed-off-by: Martin Storsjö <martin@martin.st>
The ogg decoder wasn't padding the input buffer with the appropriate
FF_INPUT_BUFFER_PADDING_SIZE bytes. Which led to uninitialized reads in
various pieces of parsing code when they thought they had more data than
they actually did.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Also, do not keep trying to find and open a decoder in try_decode_frame() if
we already tried and failed once.
Fixes always searching until max_analyze_duration in
avformat_find_stream_info() when demuxing codecs without a decoder.
Also, do not give AVCodecContext.frame_size priority for muxing.
Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
using the packet size and average bit rate.
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
For encoding, frame_size is not a reliable indicator of packet duration.
Also, we don't want to have to force the demuxer to find frame_size for
stream copy to work.
Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
Make packet buffer a parameter, don't hardcode it to be
AVFormatContext.packet_buffer.
Also move the function higher in the file, since it will be called from
read_frame_internal().
The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
This allows it to be used with get_bits without the thread of overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.
The fields "Number of Bytes" and "Number of Frames" are mixed up. "Bytes"
come first, "Frames" behind.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
This makes the packetization spec compliant for cases where one single
GOB doesn't fit into an RTP packet.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes cases where the user had specified one desired MTU
via an option, and the protocol indicates another one.
Signed-off-by: Martin Storsjö <martin@martin.st>
Frame sizes in ID3v2.3 are not synchsafe, they are simply 32be numbers.
In practice this bug is not noticeable unless the frame size takes more
than 7 bits (which is almost never for text frames).
Seeking back on EOF will reset the EOF flag, causing us to re-enter
the loop to find the next marker in the ASF file, thus potentially
causing an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Provide a way to wrap around the segment index so pseudostreaming
live through a web server and html5 browser is simpler.
Also ensure that 0 (disable) is a valid value across the options
providing wrap around.
By validating the index entries while reading, we don't need to
seek at startup to validate the entries. If the error in the
index entries is not pointing to (our definition of) the start
of packets, and there is an index entry pointing at some of the
first packets after the metadata, the invalid index can be discarded
almost immediately.
Signed-off-by: Martin Storsjö <martin@martin.st>
This returns 200 OK for OPTIONS requests and 501 Not Implemented
for all other requests.
Even though this doesn't do much actual handling of the requests,
it makes the code properly identify server requests as such, instead
of interpreting it as a reply to the client's request as it did
before.
Signed-off-by: Martin Storsjö <martin@martin.st>
For encoding, AVCodecContext.frame_size is the number of input samples to
send to the encoder and does not necessarily correspond directly to the
timestamps of the output packets.
This prevents certain tags with a default value assigned to them (as per
the EBML syntax elements) from ever being assigned a NULL value. Other
parts of the code rely on these being non-NULL (i.e. they don't check for
NULL before e.g. using the string in strcmp() or similar), and thus in
effect this prevents crashes when reading of such specific tags fails,
either because of low memory or because of targeted file corruption.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
compute_pkt_fields() is for unreliable estimates or guessing. The
keyframe information from the parser is (at least in theory) reliable,
so it should be used even when the other guessing is disabled with the
AVFMT_FLAG_NOFILLIN flag.
Therefore, move setting the packet keyframe flag based on parser
information from compute_pkt_fields() to read_frame_internal().
fixes a memleak for Vorbis and Theora, where the comment header from
avpriv_split_xiph_headers() is replaced by a buffer that must be freed
separately.
This allows opting for a lower MTU than what the AVIOContext
indicated, and allows writing into outputs that don't indicate
an MTU at all (such as plain files, which is useful for testing).
This also allows querying for the MTU via the avoption.
Signed-off-by: Martin Storsjö <martin@martin.st>
According to newer RFCs, this packetization scheme should only
be used for interfacing with legacy systems.
Implementing this packetization mode properly requires parsing
the full H263 bitstream to find macroblock boundaries (and knowing
their macroblock and gob numbers and motion vector predictors).
This implementation tries to look for GOB headers (which
can be inserted by using -ps <small number>), but if the GOBs
aren't small enough to fit into the MTU, the packetizer blindly
splits packets at any offset and claims it to be a GOB boundary
(by using Mode A from the RFC). While not correct, this seems
to work with some receivers.
Signed-off-by: Martin Storsjö <martin@martin.st>
Specifically, prevent jumping back in the file for the next index, since
this can lead to infinite loops where we jump between indexes referring
to each other, and don't read indexes that don't fit in the file.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
It is not supposed to be done outside lavc.
This is basically a revert of 818062f2f3.
It is unclear what issue this was supposed to fix, if it reappears again
it will have to be fixed in a more proper place.
The wtv-demux test change is because the sample starts with a B-frame.
We read sub_packet_h / 2 packets per line of data (during deinterleaving),
which equals zero if sub_packet_h <= 1, thus causing us to not read any
data, leading to an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This allows writing QuickTime-compatible fragmented mp4 (with
a non-empty moov atom) to a non-seekable output.
This buffers the mdat for the initial fragment just as it does
for all normal fragments, too. Previously, the resulting
atom structure was mdat,moov, moof,mdat ..., while it now
is moov,mdat, moof,mdat.
Signed-off-by: Martin Storsjö <martin@martin.st>
In nonseekable files, we already stop parsing the toplevel atoms
after finding moov and one mdat. In large seekable files (or files
that are seekable, but slowly, e.g. http), reading all the fragments
at the start can take a considerable amount of time. This allows
opting out from this behaviour.
Signed-off-by: Martin Storsjö <martin@martin.st>
If parsing moov+mdat in a non-seekable file, we currently
abort parsing directly after parsing the header of the mdat
atom. If we want to continue parsing later (if looking to
parse later fragments), we need to skip past the content of the
mdat atom, otherwise we end up parsing the content of the mdat
atom as root level atoms.
Signed-off-by: Martin Storsjö <martin@martin.st>
For video, mark the first sample in a trun which doesn't have the
sample-is-non-sync-sample flag set as a keyframe.
In particular, the "sample does not depend on other samples" flag
isn't enough to make it a keyframe, since later frames still can
reference frames prior to that one (the flag only says that that
particular frame doesn't depend on other frames).
This fixes bug 215.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids reading any old data in the AVIOContext buffer after
the seek, and indicates to the mpegts demuxer that we've seeked,
avoiding continuity check errors.
Signed-off-by: Martin Storsjö <martin@martin.st>
Enhance seeking by demuxing until the requested timestamp is
reached within the segment selected by the seek code using the
playlist info.
Some mpegts streams don't have dts set for all packets though,
this seeking method doesn't work well for that case.
Signed-off-by: Martin Storsjö <martin@martin.st>
This prevents failed assertions further down in the packet processing
where we require non-negative values for packet_size_left.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
H263 in RTP can be packetized in two formats (RFC 2190, RFC
2429/4629). The former normally uses the static payload type 34,
while the latter normally uses dynamic payload types with the
SDP format names H263-1998 or H263-2000.
Look for packets that don't look like proper RFC 2190 packets and
switch to depacketizing them according to the new format if they
match some heuristic criteria.
Signed-off-by: Martin Storsjö <martin@martin.st>
According to 14496-12, the duration should be all 1s if
the duration is unknown. This is the case if writing a moov
atom without any samples described in it (e.g. as in ismv files).
Signed-off-by: Martin Storsjö <martin@martin.st>
According to unofficial documentation, the video rate is locked to the audio
sample rate. This results in proper synchronization of audio and video
timestamps from the demuxer. This only works if the first audio packet occurs
before the first video packet or the audio sample rate is the default rate of
11111 Hz, both of which are true for all samples in our archive.
This avoids initializing a stream with dummy values or when the file does not
contain audio.
Also set duration for audio packets, using the sample rate as the time base.
Update FATE reference to account for now non-existent palette packet.
This also fixes the FATE test if frame data is not initialized in
get_buffer(), so update comment in avconv accordingly.
The rtp demuxer which listens for RTP packets and detects the
RTP payload type will currently get confused if the first packet
received is an RTCP packet. Thus ignore such packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Prefix the functions/tables brktimegm, pcm_read_seek,
dv_offset_reset, voc_get_packet, codec_movaudio_tags,
codec_movvideo_tags.
After this, lavf has no global symbols without the proper prefix.
Signed-off-by: Martin Storsjö <martin@martin.st>
Keep the old protocol name around for backwards compatibility
until the next bump.
Deprecate the method of implicitly assuming the nested protocol.
For applehttp://server/path, it might have felt logical, but
supporting hls://server/path isn't quite as intuitive. Therefore
only support hls+http://server/path from now on.
Using this protocol at all is discouraged, since the hls demuxer
is more complete and fits into the architecture better. There
have been cases where the protocol implementation worked better
than the demuxer, but this should no longer be the case.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this demuxer was created, there didn't seem to be any
consensus of a common short name for this protocol. Now
the consensus seems to be to call it hls.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is different from the "modern" RTP payload formats for H263
as defined by RFC 4629, 2429 and 3555. According to the newer RFCs,
this old one is to be considered deprecated and only be used for
interoperating with legacy systems.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows easily differentiating between both implementations within the build
system and combining the native implementation for plain RTMP with librtmp for
the RTMPE, RTMPS, RTMPT, RTMPTE protocol variants.
If muxing into mpegts, 4 byte startcodes for the first NAL
of an access unit is required. Thus it is simplest for the
RTP depacketizer to just use 4 byte startcodes everywhere.
Signed-off-by: Martin Storsjö <martin@martin.st>
When segmenting the output from the mpegts muxer, one can
now set this option when cutting to a new segment, to make sure
the next segment starts with PAT/PMT/SDT.
Signed-off-by: Martin Storsjö <martin@martin.st>
The last entry is the total size of the essence container.
Previously a TemporalOffset error would be logged, even though
segments like these are expected.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
These are common with audio atoms. Without this the demuxer would read two
bytes at a time for a mono 16-bit file.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Specially crafted files can lead the parsing code to take too long.
We fix a lot of these problems by not allowing local tags to extend
past the end of the set and not allowing other KLVs to be read past
the end of themselves.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This can happen if an index table segment has a very large IndexStartPosition.
zzuf3.mxf is an example of such a file.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Only the OPAtom demuxing logic is guaranteed to have index tables,
meaning OP1a files that lack an index would cause SIGSEGV.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Calculates based on total file size and wavetaillength from the header.
Falls back to multiplying finalframeblocks by 8 instead of 4 so that it will
at least be overestimating for 24-bit. Currently it can underestimate the
final packet size, leading to decoding errors.
Right now those muxers use the default timebase in all cases(1/90000).
This patch avoid unnecessary rescaling and makes the printed timestamps
more readable.
Also, extend the printed information to include the timebases and packet
pts/duration and align the columns.
Obviously changes the results of all fate tests which use those two
muxers.
Normally, the actual payload data contains sequence headers, too,
and the parser can extract this and set it as extradata. However,
the data in the dvc1 atom is the "official" extradata for the file.
This is required for proper stream copy of vc1 from ismv to ismv.
Signed-off-by: Martin Storsjö <martin@martin.st>
Do not use AVStream's duration for dts generation since it contains in
some cases the duration of the whole file instead of duration of the
samples in the moov. This happens if the mdhd holds the duration of the
whole file but has no entries or a zero duration in its stts.
dv: Fix null pointer dereference due to ach=0
Fixes part2 of CVE-2011-3929
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
dv: check stype
Fixes part1 of CVE-2011-3929
Possibly fixes part of CVE-2011-3936
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
This way, if the AVCodecContext is allocated for a specific codec, the
caller doesn't need to store this codec separately and then pass it
again to avcodec_open2().
It also allows to set codec private options using av_opt_set_* before
opening the codec.
It allows to check whether an AVCodecContext is open in a documented
way. Right now the undocumented way this check is done in lavf/lavc is
by checking whether AVCodecContext.codec is NULL. However it's desirable
to be able to set AVCodecContext.codec before avcodec_open2().
This makes the first packet of a track fragment run to get
the keyframe flag set properly if sample_degradation_priority
is nonzero.
This makes the keyframes flag be set properly for ismv files
created by Microsoft.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, we've only passed the key string on to the recursive
amf_parse_object for the mixedarray type, not for 'object'. By
passing the key string on, the recursive amf_parse_object can
store the amf objects as metadata.
This kind of data was seen in data from XSplit Broadcaster, received
over RTMP via Wowza. This patch allows reading this metadata.
Signed-off-by: Martin Storsjö <martin@martin.st>
Check results for av_malloc() and fix an overflow in one call.
Related to CVE-2011-3940.
Based in part on work from Michael Niedermayer.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Fixes CVE-2011-3940 (Out of bounds read resulting in out of bounds write)
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5c011706bc)
Signed-off-by: Alex Converse <alex.converse@gmail.com>
-vbsf doesn't exist anymore. It got renamed to -bsf somewhere along the
line. Update print statement accordingly.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Current demuxer recognizes several colorspace formats that begin with 'C420'
but does not yet recognize plain 'C420'. GStreamer's y4menc component
generates .y4m files with a 'C420' colorspace. This new comparison is
placed after the other 'C420' checks so that it doesn't interfere with
them.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
In this mode, no seeks will be done except for within moov/moof
fragments, which should fit within the AVIOContext buffer.
This allows pushing live smooth streaming format data to
a live publishing point on IIS over http.
Signed-off-by: Martin Storsjö <martin@martin.st>
If no data was seen for a stream decoder are returning 0 when fed with
empty packets for flushing. We can stop flushing when the decoder does
not return delayed delayed frames anymore. Changes try_decode_frame()
return value to got_picture or negative error.
CC: libav-stable@libav.org
The field frame_size isn't written to the output anywhere except
than in mov.
This facilitates stream copy from formats that don't set frame_size.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add some space around operators and wrap a comment
that extends past the 80 char "limit"/guideline.
Signed-off-by: Martin Storsjö <martin@martin.st>
It can also optionally split the file into individual fragments,
which allows it to be served from any web server without any
server side support.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes calculation of trackDuration if the MOVIentry array
is cleared. This is required by the fragmentation support in the
next patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we
do not use delta entries or slices, only StreamOffsets. OPAtom seeking
basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and
Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
Changed av_calloc to av_mallocz, added overflow checks.
It is a really bad idea to assign a video codec id
when we have set codec_type to audio and vice versa.
Prevents detection of mp2 in mxf as mpeg2video.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Specifically, this means parsing as before until we run into essence.
At that point we seek to the footer and parse until EOF. After that we start
seeking backward to the previous partition and parse that until we run into
essence or the next partition. This procedure is repeated until we encounter
the last partition we parsed in the forward direction.
The end result of all this is that large essence containers are not needlessly
parsed. This speeds up parsing large files a lot.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes 0001GL.MXF.V1.mxf_opatom.mxf and 0001GL00.MXF.A1.mxf_opatom.mxf
getting two streams each due to both using the same SourcePackageID.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Based on patch from Tomas Härdin <tomas.hardin@codemill.se>
and work by Georg Lippitsch <georg.lippitsch@gmx.at>
Changed av_calloc to av_mallocz and added overflow checks.
This fixes reading of partition packs. The code stops reading after the
operational pattern and should skip the array of essence container
labels that follow.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This avoids (for all practical cases) the issue of reusing
the same UDP port as for an earlier connection. If the remote
doesn't know the previous session was closed, he might keep
on sending packets to that port. If we always start off trying
to open the same UDP port, we might get those packets intermixed
with the new ones.
This is occasionally an issue when testing RTSP stuff with
DSS, perhaps also with other servers.
Signed-off-by: Martin Storsjö <martin@martin.st>
This check isn't relevant in the way the code currently works.
Also change a case of if (x == 0) into if (!x).
Signed-off-by: Martin Storsjö <martin@martin.st>
The s->ssrc field is the sender's SSRC, we use ssrc + 1 to get
a collision free "unique" SSRC for ourselves in the RR part.
The SDES block in the RTCP packet should describe ourselves,
not the sender.
This was fixed for the RR part in 952139a322, but wasn't
fixed for the SDES part until now.
This could cause some Axis cameras to send RTCP BYE packets
to us due to the SSRC collision.
Signed-off-by: Martin Storsjö <martin@martin.st>
Originally, sizeof(struct MOVIentry) was 48, after the reordering,
it is 40 in my build configuration.
When writing really long mov/mp4 files, this can make a difference
- this saves a bit over 2 MB of memory per hour of video (down to
10.3 MB per hour from 12.3 MB per hour initially) for a video with
75 packets per second - 25 fps + 50 audio packets (which is the
case for AMR audio).
Signed-off-by: Martin Storsjö <martin@martin.st>
The H.264 decoder needs SPS and PPS for initialization during
multi-threaded decoding. When probed single-threaded SPS and PPS are
copied to extradata and are available for proper initialization of
the decoder before the first frame is decoded.
This isn't used in practice anywhere within libav at the moment,
but change it for consistency until it is removed.
URL_RDONLY/WRONLY were fixed in commit 5b81e29593 (after the
values that actually were used were changed at the major bump,
in commit cbea3ac8), but this flag was unintentionally left unfixed.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the creation time is stored in the file as a zero, the
mov demuxer skips exporting the creation time. Currently,
files muxed without a creation time get demuxed with a
Jan 1st 1970 creation timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
rtsp.h relies on network.h and the latter conditionally defines fallback OS
structures that rely on configure tests, which are only run if networking
is enabled.
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The implicit network initialization is set to be removed in the
future, but is kept for compatibility. By not doing the implicit
initialization for non-network protocols, we avoid the warning
about avformat_network_init() not being called for these, where
it really doesn't make much sense.
Signed-off-by: Martin Storsjö <martin@martin.st>
This definition is in two files, since the definitions will move
to the private header at the next bump.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes an invalid free() with ass in avi. The sample in bug 98 passes
parts of AVPacket.data as buffer for the AVIOContext. Since the packet
is quite large fill_buffer tries to reallocate the buffer before doing
nothing. Fixes bug 98.
The fate-h264-bsf-mp4toannexb failures were caused by an integer
overflow of the unneeded multiplication.
Inspired by patch by: Michael Niedermayer <michaelni@gmx.at>
According to draft-pantos-http-live-streaming-07, 6.3.4,
the duration of the last media segment in the playlist
should be used as initial minimum reload delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the current default PES packet size, and very small audio bitrates,
audio packet duration gets too long. For players, which wait for a whole
audio packet (or more) it takes a very long time to start playing sound.
For 24kbps audio, one PES packet is about 1 second long. On Motorola STBs,
we observe about 3 second delay before the playback starts with the
default setting.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Do not assume the audio packets being always smaller than
DEFAULT_PES_PAYLOAD_SIZE.
Signed-off-by: Jindřich Makovička <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The 'fiel' atoms can be found in H.264 tracks clobbering the extradata.
MJPEG supports non field based extradata, and this data should be
preserved when copying.
This fixes demuxing of file where the first packet is not audio. Such files
are generated by our idroq muxer. It also fixes demuxing of audio only
idroq files.
Compared to just overwriting the old extradata, this has the
advantage of letting the decoder know exactly when the
extradata changed (otherwise it is changed immediately when the
new extradata packet is demuxed, even if there's old queued packets
awaiting to be decoded). This makes it easier for decoders to
actually react to the change, so they won't have to inspect
the extradata for each packet to see if it might have changed.
This works when sequentially playing a file with sample rate
changes, but if seeking past a new extradata packet in the
file, it obviously doesn't work properly. That case doesn't
work in flash player either, so it's probably ok not to handle
it.
Signed-off-by: Martin Storsjö <martin@martin.st>
Support Main Profile at High 1440 Level in MXF container,
using essence coding label from SMPTE RDD 9, table 6.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Audio header information might get scrambled and would not parse,
yet wsqva_read_packet would try to parse audio packets causing
segfaults such as floating point exception.
Fixes bugzilla #141.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids a segfault if the probe function wasn't able to
determine the format.
The bug was found by Panagiotis H.M. Issaris.
Signed-off-by: Martin Storsjö <martin@martin.st>
v410 is a packed 10-bit 4:4:4 YCbCr format used in
QuickTime.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
It sets the supplied AVFormatContext pointer to NULL after freeing it,
which is safer and its name is consistent with other lavf functions.
Also deprecate av_close_input_file().
The existing functions defined in intfloat_readwrite.[ch] are
both slow and incorrect (infinities are not handled).
This introduces a new header with fast, inline conversion
functions using direct union punning assuming an IEEE-754
system, an assumption already made throughout the code.
The one use of Intel/Motorola extended 80-bit format is
replaced by simpler code sufficient under the present
constraints (positive normal values).
The old functions are marked deprecated and retained for
compatibility.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If the sdp is generated before the rtp muxer is initialized
(e.g. as when called from the rtsp muxer), this has to be done,
otherwise the rtp muxer doesn't know that the input really is
in mp4 format.
Signed-off-by: Martin Storsjö <martin@martin.st>
If an annex b bitstream is muxed into mov, the actual written
sample is reformatted to mp4 syntax before writing.
Currently, the RTP hints that copy data from the normal video
track, where the payload data might be offset compared to the
original sample that the RTP hinting used (when 3 byte
annex b startcodes have been converted into 4 byte mp4 format
startcodes).
Signed-off-by: Martin Storsjö <martin@martin.st>
This implements reading the tag in the demuxer and adds support for writing it
in the muxer. Some example channel layout tables for muxing are included for
ac3, aac, and alac, but they are not utilized yet.
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
704af3e29c broke publishing
of rtmp streams, at least publishing to Wowza servers.
This changes all invoke commands to use nb_invokes.
Signed-off-by: Martin Storsjö <martin@martin.st>
malloc() is allowed to return NULL when zero is the argument. This
causes us to think malloc has failed and return AVERROR(ENOMEM). In
addition OS X malloc() returns an unfreeable non-NULL pointer for size
zero when alignment is greater than 16.
Pass the correct size in bits to mpeg4audio_get_config and add a flag
to disable parsing of the sync extension when the size is not known.
Latm with AudioMuxVersion 0 does not specify the size of the audio
specific config. Data after the audio specific config can be
misinterpreted as sync extension resulting in random and wrong configs.
Commit 035af99 made avconv always call an encoder when using the
null muxer. While useful for 2-pass encodes, it inadvertently
caused an extra memcpy of raw frames when decoding only.
This hack restores the old behaviour when only decoding while
allowing use of the null muxer with encoded streams as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The media_type_mask is initialized via AVOptions for the
rtsp and sdp demuxers, but it isn't available as an option
for the rtp guessing demuxer (since it doesn't really make
sense there). Therefore, it must be manually initialized
instead, since a zero value means no media types at all
are accepted.
Signed-off-by: Martin Storsjö <martin@martin.st>
For the standardized 8 kHz sample rate, this works exactly the same.
For nonstandard sample rates, the different predefined G726
names (G726-16, G726-24, G726-32, G726-40) are interpreted as an
indication of the bits per coded sample, even though their
actual bitrates aren't what the name specifies.
This feels more sane than using free-form names for nonstandard
sample rate/bitrate combinations, e.g like G726-22, G726-33
for 11025 Hz.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids overflow if frame_size is over 2147, since both
frame_size and AV_TIME_BASE are plain integers.
Signed-off-by: Martin Storsjö <martin@martin.st>
Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This simplifies the open functions by avoiding one function
call that needs error checking, reducing the amount of
extra bulk code.
Signed-off-by: Martin Storsjö <martin@martin.st>
