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>
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.
This makes the decoder output have double the magnitude
compared to before.
The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
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 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>
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>
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>
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 is required when stream copying VC1 in ismv - there's one
global header in the moov atom, but keyframes have a separate
sequence header prepended.
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>
This fixes crashes on exit when closing a bitstream filter that
hasn't allocated any private data, on OS X.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
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>
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>
Earlier, calling avcodec_encode_audio worked fine even if time_base
wasn't set. Now it crashes due to trying to scale the output pts to
the codec context time base. This affects e.g. VLC.
If no time_base is set for audio codecs, set it to the sample
rate.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
MinGW doesn't have sleep, only _sleep (which is deprecated),
Sleep (which is defined in winbase.h and not in the standard
C headers) and usleep.
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>
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>
Martin Storsjö