pkgsrc - [no description]

Age	Commit message (Collapse)	Author	Files	Lines
2010-01-20	Bump PKGREVISION for gupnp/gssdp API changes.	wiz	1	-1/+2

2010-01-20	Update to 0.0.17:	wiz	1	-2/+2
	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
2009-10-31	Update to 0.0.16:	wiz	1	-3/+2
	Version 0.0.16 rtpspecialsource: Remove want_source() method get_codec() function does the same thing rtpdtmfsoundsource: Implement get_codec method rtpdtmfeventsource: Implement get_codec method rtpspecialsource: Add new get_codec method rtp: Check if the codec changed when removing special sources rtp: Allow checking if a codec is valid for sending even if it has no way to build a codecbin rtpcodecnego: Fix doc string rtpspecialsource: Move static function closer to its use place rtpspecialsource: Fix over-80 line rtpsession: Check/update secondary sources even if the primary one doesn't change tests: Tests changing the dtmf PT mid-call tests: Make sure dtmf events are really received test: Test changing the dtmf_id tests: dtmf method is not always auto rtpsession: Only emit send-codec-changed message after the special codecs have been changed rtpsession: Don't leak iterator on linking failure rtpsession: Cleanup send codecbin on failure rtpsession: Print error on session dispose problems rtpdtmfsoundsource: Correctly check the presence of elements rawudp: Use %d for ints, not %s configure: quiet automake portability bs msnstream: Make send sink async=false for now msnstream: Don't keep lock into set_remote_candidates tests: Test invalid property name in fs_element_added_notifier_from_keyfile element-added-notifier: Don't crash on invalid property rtpconference: Don't assert on non-existing sdes parts rtpspecialsource: Dispose is not always called twice, cleanup in finalize rtpsession: Remove useless ref Version 0.0.15.1 Version 0.0.15 Require gst-p-bad 0.10.14 for mimic tests: Unlock src before setting it to playing tests: Refrain from using the thread unsafe version of failure in the nice test rtpsession: Keep ref on stream while associating substreams to it rtpsubstream: Remove another double-unlock in error case rtpsession: Don't double-unlock rtpsession: Fix leaking caps on signals after dispose rtpsession: Fix potential leak if already disposed rtpsubstrea: Remove unused variable elementaddednotifier: Use g_connect_signal_object Otherwise each element had a ref on the notifier and relied on the not thread safe weak references. rawudp: Emit local candidates if there are no local interfaces suitable for UPnP rawudp: Add some UPnP debug messages glib-gen: Use single = instead of == for portability msnconnection: Check return values from recv() msnsession: Conference must always set before get_property msnsession: Only try to lock conference if it has been set rtpsession: Initialise variable to NULL Makes coverity happy msnconnection: Remove unused variables rtpstream: Correct documentation rtpsession: Unref transmitter src/sink in dispose Unref element from g_object_get(), fixes leak elementaddednotifier: Unref element in iterator loop Fixes leak elementadded: Use gst_value_deserialize to read properties Use the existing function instead of having our own less-capable re-implementation Version 0.0.14.1
2009-09-23	Remove "PYTHON_VERSIONS_ACCEPTED= 26 25 24" which is unnecessary	tron	1	-2/+1
	after Python 2.3 has been removed from "pkgsrc". Approved by Thomas Klausner.
2009-08-26	bump revision because of graphics/jpeg update	sno	1	-1/+2

2009-08-17	Initial import of farsight2-0.0.14:	wiz	1	-0/+47
	The Farsight project is an effort to create a framework to deal with all known audio/video conferencing protocols. On one side it offers a generic API that makes it possible to write plugins for different streaming protocols, on the other side it offers an API for clients to use those plugins. The main target clients for Farsight are Instant Messaging applications. These applications should be able to use Farsight for all their Audio/Video conferencing needs without having to worry about any of the lower level streaming and NAT traversal issues. Farsight forms an integral part of the Telepathy framework. It is used by Empathy through the Telepathy-Farsight library. It can also be easily used on embedded platforms by using Stream-Engine. The Telepathy-Farsight library binds it to the Connection Managers via D-Bus and the Telepathy Media Stream Spec and is used for all their streaming requirements.