| Commit message (Collapse) | Author | Age | Files | Lines |
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Patch by Josh Allmann (joshua allmann gmail com)
Originally committed as revision 22636 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 22109 to svn://svn.ffmpeg.org/ffmpeg/trunk
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but doesn't actually do that. What's worse, it creates timestamp adjustments
that are different per stream within a session, leading to a/v sync issues.
See discussion in thread "[FFmpeg-devel] rtp streaming x264+audio issues (and
some ideas to fix them)". Patch suggested by Luca Abeni <lucabe72 email it>.
Originally committed as revision 21857 to svn://svn.ffmpeg.org/ffmpeg/trunk
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what e.g. RealPlayer does. This allows proper port forwarding setup in NAT-
based environments.
Patch by Martin Storsjö <$firstname at $firstname dot st>.
Originally committed as revision 21856 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 21740 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Patch by Alexis Ballier, alexis D ballier A gmail
Originally committed as revision 21601 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 21512 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 20916 to svn://svn.ffmpeg.org/ffmpeg/trunk
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qualification task, see "RTP/Vorbis payload implementation (GSoC qual
task)" thread on mailinglist.
Originally committed as revision 18509 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 18494 to svn://svn.ffmpeg.org/ffmpeg/trunk
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SDP line handler that parses the streamID in the SDP so that ASF stream
data can be matched to their respective streams in the RTSP demuxer. See
"[PATCH] RTSP-MS 12/15: ASF payload support" thread on mailinglist.
Originally committed as revision 18061 to svn://svn.ffmpeg.org/ffmpeg/trunk
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payload handlers take care of that themselves at their own option. What this
patch really does is "fix" a bug in MS-RTSP protocol where incoming packets
are always coming in over the connection (UDP) or interleave-id (TCP) of
the stream-id of the first ASF packet in the RTP packet. However, RTP packets
may contain multiple ASF packets (and usually do, from what I can see), and
therefore this leads to playback bugs. The intended stream-id per ASF packet
is given in the respective ASF packet header. The ASF demuxer will correctly
read this and set pkt->stream_index, but since the "stream" parameter can
not be known to rtpdec.c or any of the RTP/RTSP code, the "st" parameter
in all these functions is basically invalid. Therefore, using st->id as
pkt->stream_index leads to various playback bugs. The result of this patch
is that pkt->stream_index is left untouched for RTP/ASF (and possibly for
other payloads that have similar behaviour).
The patch was discussed in the "[PATCH] rtpdec.c: don't overwrite
pkt->stream_index in finalize_packet()" thread on the mailinglist.
Originally committed as revision 17767 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 17765 to svn://svn.ffmpeg.org/ffmpeg/trunk
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called for all packets with an internal handler function but only for
non-first packets from dynamic payload parse_packet() handlers. This patch
fixes that. Bug was noticed by Luca in "[PATCH] rtpdec.c: don't overwrite
pkt->stream_index in finalize_packet()" thread.
Originally committed as revision 17764 to svn://svn.ffmpeg.org/ffmpeg/trunk
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under review. See "[FFmpeg-devel] RTP mark bit not passed to parse_packet"
thread on mailinglist.
Originally committed as revision 17616 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 17016 to svn://svn.ffmpeg.org/ffmpeg/trunk
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to the parse_packet() function pointer in RTPDynamicProtocolHandlers. This
allows these functions to peek back and retrieve values from the demuxer's
context (or RTSPState). The ASF/RTP payload parser will use this to be able
to parse SDP values (which occur even before the payload ID is given), store
them in the RTSPState and then retrieve them while parsing payload data. See
"[PATCH] RTSP-MS 13/15: add RTSP demuxer AVFormatContext to parse_packet()
function pointer (was: transport context)" mailinglist thread.
Originally committed as revision 17015 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 16817 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 16684 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 16115 to svn://svn.ffmpeg.org/ffmpeg/trunk
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(and thus preparing for the introduction of RDTDemuxContext) in a next patch.
See discussion in "RDT/Realmedia patches #2" thread on ML.
Originally committed as revision 15542 to svn://svn.ffmpeg.org/ffmpeg/trunk
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not use RTPDemuxContext, but rather take a pointer to the payload context
directly. This allows using payload handlers regardless over the transport
over which they were sent, and prepares for the introduction of a future
RDTDemuxContext. See discussion in "RDT/Realmedia patches #2" thread on ML.
Originally committed as revision 15541 to svn://svn.ffmpeg.org/ffmpeg/trunk
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The feature_tests.h header from Sun systems (Solaris/OpenSolaris) will abort
the build if _XOPEN_SOURCE is defined to 500, and C99 is requested (as well
as POSIX.1-2001), and will only accept it to be defined to 600.
inspired by a patch from Diego Pettenò, flameeyes gmail com
Originally committed as revision 15460 to svn://svn.ffmpeg.org/ffmpeg/trunk
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it so that I can use it in rdt.c as well. See discussion in "Realmedia patch"
thread on ML.
Originally committed as revision 15233 to svn://svn.ffmpeg.org/ffmpeg/trunk
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(fix a bug in the RTP demuxer)
Originally committed as revision 14909 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 14766 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 14211 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 14046 to svn://svn.ffmpeg.org/ffmpeg/trunk
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(correctly compute the presentation times based on the RTP timestamps
and the RTCP SR packets)
Originally committed as revision 14045 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 13098 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 12449 to svn://svn.ffmpeg.org/ffmpeg/trunk
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This can be used later by RDT to get the flags from the RTP packet and
use that for the RealMedia packet (such as whether this RTP packet
represents a keyframe or not). For discussion, see "[PATCH] Realmedia
/ RTSP (RDT)".
Originally committed as revision 11557 to svn://svn.ffmpeg.org/ffmpeg/trunk
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see "[PATCH] Realmedia / RTSP (RDT)" thread on ML.
Originally committed as revision 11494 to svn://svn.ffmpeg.org/ffmpeg/trunk
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if there is one. See "[PATCH] Realmedia / RTSP (RDT)" thread on ML.
Originally committed as revision 11493 to svn://svn.ffmpeg.org/ffmpeg/trunk
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dependencies
Originally committed as revision 11406 to svn://svn.ffmpeg.org/ffmpeg/trunk
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