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>
