wiz ad36134bab Update to 0.0.17: ... Version 0.0.17 tests: Add test for telephone-event events parameter nego rtpspecificnego: Add handling of telephone-event event ranges tests: Skip tests if no local candidates are produced rtcpfilter: Reduce the packet size when reducing the packet tests: Skip libnice tests if it finds no local candidates rtpdtmfsoundsource: Respect the ptime/maxptime too tests: Add test ptime/maxptime passing rtpsession: Set the ptime/maxptime on the send codec bin caps rtpcodecnego: Negotiate the ptime/maxptime rtpconference: Add function to make gst caps while keeping the ptime rtpcodecnego: Add function to copy the list of codecs with the send-side ptime tests; Add test for fscodec ptime/maxptime handling codec: Add ptime codec: Add maxptime tests: Take rtpsession lock during message emissions This ensures that it is not held across message emissions. tests: Add debug-blocks rtpsubstream: Keep ref on substream while callbacks are invoked rtpsubstream: Put codec/codecbin inside loop rtpsubstream: Use rw-lock to make sure the substream really stops rtp: Move locking into callback rtpsubstream: Don't hold session lock too much while setting new codecbin rtpsubstream: Move modification locking to blocked function Also allow only one thread to be in substream blocked function at once. rtp: Move substream blocking logic into substream rtp: Don't include marshaller headers in headers rtp: Depend on the correct var for marshaller list generation rtcpfilter: Add gst-p-base paths to Makefile.am Patch contributed by Armijn Hemel <armijn@loohuis-consulting.nl> rawudp: Remove upnp-request-timeout, it was a terrible idea Substitute deprecated Glib symbol: g_mapped_file_free Use g_mapped_file_unref if Glib >= 2.22 is available http://bugs.freedesktop.org/show_bug.cgi?id=21422 rtpsession: Only add stream to list if its creation worked README: Require gst-p-bad 0.10.17 for dtmfsrc dtmfsrc can do do more than 8000 Hz, that has only been fixed in gst-plugins-bad 0.10.17 rtpdtmfsound: Try hardwired PCMx only if the clock-rate is 8000 rtp: Lookup codec with config is always for sending, so make it explicit Also, the dtmf sound will always get a valid codec now. rtpconference: Make message about gst_bin_add failure more accurate rtpdtmfsoundsource: Ignore codecs that don't have a blueprint tests: Test dtmf as sound tests: Make recv-pipeline per test rtpdtmfsoundsource: Use main codec if PCMA/U are not available rtpspecialsource: Make local class_get_codec function static rtp: Regroup CodecBlueprint related functions in one place rtpspecialsource: Rename negotiated_codecs to negotiated_codec_associations This way, the list contents can be guessed rtpsession: Don't need to set queue-delay anymore rtpsession: Split codecbin generation from factory from profile tests: Make it build against GUPnP 0.13 msnsession: Check if dispose has already been called fstransmitter: uint can't be < 0 rawudp: Bring upnp discovery timeout down to 2 seconds tests: Verify that it is not possible to disable all codecs Add a reserve-pt to guarantee that it is not possible to disable all codecs rtpcodecnego: Verify if there are any valid local codecs left after applying preferences rtpsession: Make error message less cryptic Version 0.0.16.1		2010-01-20 09:26:52 +00:00
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patches	Update to 0.0.16:	2009-10-31 02:47:32 +00:00
buildlink3.mk
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distinfo	Update to 0.0.17:	2010-01-20 09:26:52 +00:00
Makefile	Update to 0.0.17:	2010-01-20 09:26:52 +00:00
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