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added more info to list-cam, fixes #237
don't log junk
able to restart senders without breaking receivers
receiver doesn't quit on sender timeout, still can't handle new senders though
merged unsingleton_pipeline branch into trunk@4525
raw doesn't work with more than 8 channels, so prevent users from trying it
more refactoring of pipeline construction
merged back branches/scrap_telnet@4495 into trunk @ 4495
create our own sockets for udpsrc/sink, need to test with multicast
created sockfd branch from trunk@4449
broken commit for file move
receiver quits on sender timeout
print a message in receiver when sender quits
merged services@3600:3951 in trunk * services and streams using deferreds * start/stop should work for both alice and bob * streams configuration ala OSC * updated dia classes diagrams (doc/classes/proto_conf_v3.dia) * dropped use of IPCP telnet protocol to control milhouse...
merged fixed caps branch back into trunk
get caps from payloader instead of rtpbin's udpsink, a little cleaner, but still only works up to gst-plugins-bad-0.10.12
using boost lexical cast instead of stringstream
only print rtp stats for pipeline that is playing
refactored rtp
rtp stats now in log info form, smoke test added to telnet_thrillhouse, gtk crap out of pipeline, respecting naming conventions in audiosink
got rid of unused var
got rid of problematic catch all
proper rtp stats info
code reuse, yay
using quality (Variable bitrate) rather than bitrate for theora
assert now tassert for throw-assert
location and device are distinct parameters
rtp message posted to tcpserver
safer rtp session labelling, debugging playback ctrls added
fixed error in rtp destructors whicch would try to remove names that weren't there in certain cases and lead to segfaults/bad things, as well as more checking
fixed h263, in a manner of speaking
vorbis encoding in its own thread, better rtp messages
unified logging, could probably still use a makeover format wise
unified logging, almost
more obvious rtp stats
correct types for rtpstats
printing rtp
bitrate info yay
use offset from video and audioports for caps transmission
added firereset to trunk, updated sropulpof.py to work with api changes
rtp api improvements, latency setting added
add some jitterbuffering
get rid of rtp reporting callback when all our rtpsessions are toast
purged the evil of the drop-on-latency setting from rtpbin
trying without drop on latency
don't ever set jitterbuffer latency to 5 again
give an ever so slight jitter buffer
increasing latency doesn't seem to affect quality
try default latency again
try with jitterbuffer latency
pushing up default args, keep maugis with the times
copyright
fixed most glsink issues
printing some stuff
fixed port number check gst/factories
begins quest to remove private state variables, which should actually be queried directly from gst
unit test for sropulpof defaults
printing of rtpstats readable, but only enabled if enable-debug is set
cleanup in rtpbin
rtp stats reporting now works
rtpbin can report on even more potentially meaningless stats
video and audio caps are now sent via tcp
rtp jitter reporting in source, commented out for now
fixed jack channel mapping
Added payload types to video receiver caps until Gstreamer bug #565509 is fixed. Added compile time check for sropulpof to run it with verbosegstreamer pipeline info.
more rtp api stuff
added bandwidth get set for rtp, which has no visible effect on gstreamer
fixed h264 caps
setting rtpbin's jitterbuffer latency to avoid delay
util.h general header
took out request session
backward compatible git-commands
moved/renamed directories
