* commit 'f797b134cad4d248b1c8955659997980d0668bc3':
rtpenc_chain: Don't copy the time base to the source stream by default
See: 1fe40e1b05
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Only copy it manually in the muxers where it makes sense (rtspenc,
sapenc). Don't touch the original AVStream in movenchint, where
the original AVStream should be kept untouched.
This fixes the normal tracks in RTP hinted files after
abb810db - the hint tracks were ok while the normal media tracks
were broken, noticed by Michael Niedermayer.
This reverts abb810db but achieves the same effect for the other
muxers.
Signed-off-by: Martin Storsjö <martin@martin.st>
For muxing, it accepts
both 0 and AV_NOPTS_VALUE. For demuxing, it will present
AV_NOPTS_VALUE when start_time_realtime is unknown.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If set, and if TCP is available as RTSP RTP transport, then TCP will be
tried first as RTP transport.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
An SDP description normally only contains the target IP address
and port for the packets. This means that we don't really have
any clue where to send the RTCP RR packets - previously they're
sent to the destination IP written in the SDP (at the same port),
which rarely is the actual peer. And if the source for the packets
is on a different port than the destination, it's never correct.
With a new option, we can choose to send the packets to the
address that the latest packet on each socket arrived from.
---
Some may even argue that this should be the default - perhaps,
but I'd rather keep it optional at first. Additionally, I'm not
sure if sending RTCP RR directly back to the source is
desireable for e.g. multicast.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'f4d371b9737c0405b3bc46d7ca0c856c0a8616b1':
rtsp: Don't include the listen flag in the SDP demuxer flags
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It's only relevant for the RTSP demuxer. Similarly, the custom_io
flag is only present in the SDP demuxer options list.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'b7e6da988bfd5def40ccf3476eb8ce2f98a969a5':
rtpproto: Move rtpproto specific function declarations to a separate header
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1f57d60129b0e297cd197c6031c4439b30a6b503':
rtsp: Support RFC4570 (source specific multicast) more properly.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add support for domain names, for multiple source addresses,
for exclusions, and for session level specification of addresses.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '36fb0d02a1faa11eaee51de01fb4061ad6092af9':
rtsp: Support multicast source filters (RFC 4570)
rtpproto: Check the source IP if one single source has been specified
rtpproto: Support IGMPv3 source specific multicast inclusion
Conflicts:
libavformat/rtpproto.c
libavformat/rtsp.c
libavformat/rtsp.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This supports inclusion of one single IP address for now,
at the media level. Specifying the filter at the session level
(instead of at the media level), multiple source addresses,
exclusion, or using FQDNs instead of plain IP addresses is not
supported (yet at least).
Signed-off-by: Martin Storsjö <martin@martin.st>
Passes Source-Specific Multicast parameters read from an sdp file through to the UDP socket code,
allowing source-specific multicast streams to be correctly received. As an integral part of this
change, additional checking (currently only enabled in the case of SSM streams, but probably
useful in similar scenarios) has been added to the RTP protocol handler to distinguish UDP packets
arriving from multiple sources to the same port and process only the expected packets
(those transmitted from the expected UDP source address). This resolves an issue identified
when multiple instances of FFmpeg subscribe to different Source-Specific Multicast streams
but with each sharing the same destination port.
Signed-off-by: Edward Torbett <ed.torbett@simulation-systems.co.uk>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1dd1b2332ebbac710d8e0214cec7595e118f2105':
rtsp: Include an User-Agent header field in all requests
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b3ea76624ad1baab0b6bcc13f3f856be2f958110':
vf_aspect: use the name 's' for the pointer to the private context
Remove commented-out debug #define cruft
Conflicts:
libavcodec/4xm.c
libavcodec/dvdsubdec.c
libavcodec/ituh263dec.c
libavcodec/mpeg12.c
libavfilter/avfilter.c
libavfilter/vf_aspect.c
libavfilter/vf_fieldorder.c
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e926b5ceb1962833f0c884a328382bc2eca67aff':
avformat: Drop unnecessary ff_ name prefixes from static functions
Conflicts:
libavformat/audiointerleave.c
libavformat/mux.c
libavformat/mxfenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
lavf: Add a protocol for SRTP encryption/decryption
rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This only takes care of decrypting incoming packets; the outgoing
RTCP packets are not encrypted. This is enough for some use cases,
and signalling crypto keys for use with outgoing RTCP packets
doesn't fit as simply into the API. If the SDP demuxer is hooked
up with custom IO, the return packets can be encrypted e.g. via the
SRTP protocol.
If the SRTP keys aren't available within the SDP, the decryption
can be handled externally as well (when using custom IO).
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1':
rtsp: Remove an outdated comment
rtsp: Remove references to weirdly named variables in other files
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is unclear what the bug exactly was and if it ever was fixed,
and we don't even support decoding via faad any longer. The
comment has been present since d0deedcb in 2006.
Signed-off-by: Martin Storsjö <martin@martin.st>
One of them is renamed now, but mentioning it by name serves
no purpose here. The other table mentioned ceased to exist
under that name in 4934884a1 in 2006.
Signed-off-by: Martin Storsjö <martin@martin.st>
This sends NACK for missed packets and PLI (picture loss indication)
if a depacketizer indicates that it needs a new keyframe, according
to RFC 4585.
This is only enabled if the SDP indicated that feedback is supported
(via the AVPF or SAVPF profile names).
The feedback packets are throttled to a certain maximum interval
(currently 250 ms) to make sure the feedback packets don't eat up
too much bandwidth (which might be counterproductive). The RFC
specifies a more elaborate feedback packet scheduling.
The feedback packets are currently sent independently from normal
RTCP RR packets, which is not totally spec compliant, but works
fine in the environments I've tested it in. (RFC 5506 allows this,
but requires a SDP attribute for enabling it.)
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '8729698d50739524665090e083d1bfdf28235724':
rtsp: Recheck the reordering queue if getting a new packet
lavr: log channel conversion description for any-to-any functions
lavr: mix: reduce the mixing matrix when possible
lavr: cosmetics: reindent
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If we timed out and consumed a packet from the reordering queue,
but didn't return a packet to the caller, recheck the queue status.
Otherwise, we could end up in an infinite loop, trying to consume
a queued packet that has already been consumed.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'e96406eda4f143f101bd44372f7b2d542183000a':
rtsp: Add support for depacketizing RTP data via custom IO
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3f95f0dda55fca74b646937095a02a8fa9776622':
rtpdec: Move the URLContext used for RTCP RR out from the context, to a parameter
Merged-by: Michael Niedermayer <michaelni@gmx.at>
To use this, set sdpflags=custom_io to the sdp demuxer. During
the avformat_open_input call, the SDP is read from the AVFormatContext
AVIOContext (ctx->pb) - after the avformat_open_input call,
during the av_read_frame() calls, the same ctx->pb is used for reading
packets (and sending back RTCP RR packets).
Normally, one would use this with a read-only AVIOContext for the
SDP during the avformat_open_input call, then close that one and
replace it with a read-write one for the packets after the
avformat_open_input call has returned.
This allows using the RTP depacketizers as "pure" demuxers, without
having them tied to the libavformat network IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
