* 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>
Theoretically using start_time should also work if seeking is available and we
could determine that the next packet after a flush packet is the first packet
of a stream, but I could not think of an easy and clean way to do that, that is
why I sticked to the no seeking available condition for now.
Fixes ticket #2647.
Signed-off-by: Marton Balint <cus@passwd.hu>
Previously we estimated the audio packet pts instead of the frame pts,
therefore it only worked within a single packet (containing multiple frames).
The new method works accross seperate audio packets as well and also handles
better the case if a decoder buffers several packets before outputting a
decoded frame.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also use negative stream_index for signaling obsolete audio packets. Using the
size alone is not enough, because size is 0 for null packets as well.
Signed-off-by: Marton Balint <cus@passwd.hu>
This also fixes the case where negative chapter ids where input
And fixes the case where remuxing from mkv changed chapter ids
Found-by: Luca Barbato
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
There is no reason why this should copy the audio data in a very
complicated way. Also, strictly write the first plane, instead of
writing the whole buffer. This is more helpful in context of the
example. This way a user can clearly confirm that it works by playing
the written data as raw audio.
This assumes one audio packet is decoded one time. This is not true:
packets can be partially decoded. Then you have to "adjust" the packet
and pass the undecoded part of the packet to the decode function again.
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>
If another peer is sending unicast packets to the same port that
we are listening on, those packets can end up being received despite
using source specific multicast. For those cases, manually check the
source address of received packets against the intended source address.
This only handles the case when the source list is one single IP
address for now, which probably is the most common case.
Based on a patch by Ed Torbett.
Signed-off-by: Martin Storsjö <martin@martin.st>
Blocking/exclusion is not supported yet.
The rtp protocol parameter takes the same form as the existing
sources parameter for the udp protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavcodec/jpeg2000dec.c
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cehoyos/master:
Fix pix_fmt detection in the native jpeg2000 decoder.
lavf/concat: Never fail for sample aspect ratio 0:1.
lut3d: Fix reading 3dl files with leading comments.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
There is no record of this ever being used at all, anywhere,
since the feature was added in 2effd27446.
This gets rid of extra linker tricks just to support a feature
that isn't used, simplifying portability to other platforms.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fix "Value 0.000000 for parameter 'srch' out of range"
error message when source or destination height is 1.
Note: since the av_opt_set_int() calls are not checked for
failure and the interlaced scalers are not actually used,
this error has no consequence apart from a frightening message
in the log.
It's the official (or recommended) name for comment/description entries.
See https://www.xiph.org/vorbis/doc/v-comment.html
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some players, like foobar2000 or modern versions of WMP, create WAV
files using the ITRK tag for track instead of IPRT
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Consistently use "cpu_flags" as variable/parameter name for CPU flags
Conflicts:
libavcodec/x86/dsputil_init.c
libavcodec/x86/h264dsp_init.c
libavcodec/x86/hpeldsp_init.c
libavcodec/x86/motion_est.c
libavcodec/x86/mpegvideo.c
libavcodec/x86/proresdsp_init.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* https://github.com/lukaszmluki/ffmpeg:
ftp: warning about pure-ftp server used as and output
ftp: comments
ftp: remove unused headers
ftp: fix interrupt callback misuse
Merged-by: Michael Niedermayer <michaelni@gmx.at>
FTP protocol used interrupt callback to simulate nonblock
operation which is a misuse of this callback.
This commit make FTP protocol fully blocking and removes
invalid usage of interrutp callback
Also adds support for multiline responses delimited with dashes
