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2011-06-10recursive bump from textproc/icu shlib major bump.obache1-2/+2
2011-04-22recursive bump from gettext-lib shlib bump.obache1-1/+2
2011-03-11Update to 0.0.26:wiz1-2/+2
Version 0.0.26 rawconference: Correctly check if thread is internal rawstream: Don't start sending before having codecs rawsession: Only manipulate the valve from the session itlsef rawsession: Simplify transform bins creation rawsession: Remove g_debug rawsession: Unref the right object rawsession: Only remove sink if it has been added rtpconference: Correctly check if a thread is internal rtpstream: Fix reference leak in fs_rtp_stream_set_negotiated_codecs_unlock() Keep a ref to the fakesink fsrawconference: Make the construction more consistent In the construction of a raw session we add a bunch of elements. For all elements unref them in _constructed if adding them to the bin fails, for all other failures, leave it to the _dispose function to remove and unref the elements Use full prefix, even for private functions Add a transformation bin the source pipeline As upstream negotiation in Gstreamer still doesn't actually work, we'll need to change transform elements around every time the caps are changed as that will cause a re-negotiation and things will keep working.. Unfortunately managing dynamic pipelines has its own challenges, so add a tee ! fakesink which will eat all the errors for us... fsrawconference: Make fsrawstream explicitely ask the session to set the direction fsrawconference: Cope with fs_raw_session_new returning NULL
2011-02-21Update to 0.0.25:wiz1-4/+3
Version 0.0.25 nicestreamtrans: Fix off-by-one bug https://bugs.freedesktop.org/show_bug.cgi?id=34291 Version 0.0.24.1 Version 0.0.24 tests: Rtcp test doesn't make sense in raw rtp: add default prefs to EXTRA_DIST tests: Fix another race in tests nicestream: Skip Nice errors if the component has never been ready nicestream: Fix small leak nicestream: Sort ipv4 addresses first utils: Fix doc string utils: the keyfile stuff already checks the user dirs utils: Pass the element directly instead of its factory name utils: Check default properties/codecs in user data dir too rtp: Add default-element-properties utils: Add function to get default element properties rtp: Add default codec preferences utils: Add function to get default codec preferences raw: Don't delete non-generated files Remove the temporary socket directory after usage nicetransmitter: Place the local socket in the tmp dir Don't hardcode /tmp, instead use g_get_tmp_dir to potentialy get it from the environment, but falling back to /tmp nicetransmitter: Add documentation for create-local-candidates nicetransmitter: Add an option for the transmitter to pick the local side rawconference: This is really meant to be called on the stream. nicetrans: Only emit local-candidate after gathering Unfortunately libnice doesn't currently support doing connectivity checks untill it has finished gathering. If we send a remote peer our candidates before finishing gathering they can start sending us connectivity checks before we're ready for them... So instead sends the local candidates in one batch when gathering is finished, so we'll be ready for the connectivity checks. rawconf: Put the whole caps into the encoding_name in codecs rawconference: Make FsRawStream codec doc visible. docs: Improve the title docs: Add docs for the raw plugin raw: Remove trailing whitespace raw: Simplify session notification of new stream codecs rawstream: Simplify set_remote_codecs cuseless rawsession: Codec has already been validated raw: Don't check for stuff in the codecs that is meaningless for raw rawconference: Add a test with the shm transmitter. rawconference: Remove stream from session in stream's dispose. There's a chance that removing the stream when the session has it weak-reffed can be called from a streaming thread. This can cause it to crash and/or deadlock. This patch changes the stream to call the remove_stream function in the session in its dispose function. The stream already protects itself from being disposed in a streaming thread and therefore prevents the crash/deadlock. rawconference: Use local conference variable. tests: Split the rtpconf extra init into separate callbacks for stream and conf tests: Split the rawconf extra init into separate callbacks for stream and conf rawconference: Remove weak_ref when done. rawconference: Dispose FsRawStream in a separate thread if needed. rawconference: Add fs_raw_conference_is_internal_thread. rawconference: Fix trailing whitespace. rawconference: Correct an error message. rawconference: Wait to add the transmitter's gst-sink until sending. rawconference: Dispose of objects in a single place in new_stream. rawconference: blocking_id will always be 0 here. rawconference: Remove transmitter-pad from the public API. rawconference: Correctly use g_value_set_boxed instead of _take_boxed. rawconference: Use macro instead of g_mutex_lock directly. This patch creates and uses FS_RAW_SESSION_LOCK and _UNLOCK and FS_RAW_STREAM_LOCK and _UNLOCK to improve the ability to debug mutexes. rawconference: Add @author to the files I made. rawconference: Misc style and error checking fixes to Sjoerd's commits. When adding streams, sync the element states with the parent element When removing a stream, make the valve drop packets again rawconference: Change signature of function to avoid collision. This patch changes the signature of fs_codec_to_gst_caps to fs_raw_codec_to_gst_caps to avoid colliding with a function of the same name in the FsRtpConference plugin. rawconference: Keep reference to GstObjects in FsRawStream. rawconference: Actually store the src_pad in FsRawStream. rawconference: Remove unused member from FsRawStream private struct. rawconference: Improve locking in FsRawStream. rawconference: Simplify FsRawSession dispose a little. rawconference: Hold references to GstObjects in FsRawSession. rawconference: Improve FsRawSession's locking. rawconference: Remove elements from bin if sync_state_with_parent fails. rawconference: Simplify a little of removing streams. rawconference: Simplify FsRawSession's dispose function. rawconference: Remove redundant gst_element_sync_with_parent call. rawconference: Fix implemention of FsRawSession's current-send-codec. rawconference: Store FsRawSession codecs and notify on change. rawconference: Fix potential double-free. rawconference: Deactivate pad after removing from bin. rawconference: Remove unneeded variable and just return value. rawconference: Fix copy/paste errors. rawconference: Use correct pad template. rawconference: Fix disposed testcase. rawconference: Free transmitter src and sink when removing streams. rawconference: Set the correct error in fs_raw_session_new_stream. rawconference: Fix base test. FsRawConference doesn't generate codecs. rawconference: Use optional_parameters for codec properties. rawconference: Abstract converting FsCodec to GstCaps. rawconference: Add tests for FsRawConference plugin. This patch adds tests for the FsRawConference plugin. Virtually all of the code is from the FsRtpConference plugin testsuite. rawconference: Add data probe and src_pad_added emission. rawconference: Set capsfilter caps when set_remote_codecs is called. rawconference: Plug memory leak. rawconference: Set initial valve drop settings after creation. rawconference: Set ST's "sending" property when setting "direction". rawconference: Set booleans instead of bitmasked integers. rawconference: Fix some GstElement refcount issues. rawconference: Implement FsRawSession's remote codec handler. rawconference: Implement FsRawSession's codecs properties. Implement the FsRawSession's "codecs" and "codecs-without-config" properties. rawconference: Link the FsRawSession's capsfilter and transmitter_sink. rawconference: Free the FsRawSession's FsTransmitter. rawconference: Add to FsConference and partially link transmitter. rawconference: Fix getting an out of range warning on a gboolean value. rawconference: Fix some type issues in fs_raw_session_new_stream. rawconference: Improve setting the direction. rawconference: Implement the remote-codecs FsRawStream property. rawconference: Implement fs_raw_stream_set_remote_codecs. rawconference: Create and connect FsStreamTransmitter signal handlers. rawconference: Implement fs_raw_stream_set_remote_candidates. rawconference: Remove fs_raw_stream_set_tos_locked. rawconference: Add FsStreamTransmitter. rawconference: Implement fs_raw_session_get_stream_transmitter_type. rawconference: Add FsTransmitter member. rawconference: Add FsRawStream class files. rawconference: Add capsfilter to the session pipeline. rawconference: Add an id to FsRawSessions and support creating them. rawconference: Implement fs_raw_conference_list_transmitters. rawconference: Add the FsRawSession class. These files have been copied directly from the FsMsnSession class and have simply been renamed. More modifications will be needed. P.S. The section documentation has also been altered to better suit the FsRawSession class. rawconference: Remove cname from FsRawParticipant. rawconference: Add FsRawParticipant. rawconference: Add base FsRawConference class and plugin structure. Version 0.0.23.1 Version 0.0.23 common-modified: Dist another stamp file nice: Update to use the nice 0.1.0 API nice: Add compatibility for MS Office Communicator 2007 R2 example gui: Keep a ref to the FsElementAddedNotifier to keep it alive example gui: Set the necessary properties for x264enc rtpsession: Really fix dispose checking rtpsession: Only set disposed to TRUE when actually disposing tests: Add a test of codecs-ready before calling any method Make sure the codecs-ready is not TRUE if no methods have been called yet and some codecs that require discovered parameters are missing. rtpsession: Make sure the original codecs are propertly setup Do the update codecs when creating a FsSession so that original codecs have the required bits for the parameter gathering. tests: Add test for pad alloc in fsfunnel Patch by Yongnian Le <yongnian.le@intel.com> funnel: Implement pad allocation Patch by Yongnian Le <yongnian.le@intel.com> https://bugs.freedesktop.org/show_bug.cgi?id=32208 Use portable 'g_snprintf' instead of 'snprintf' https://bugs.freedesktop.org/show_bug.cgi?id=32276 Replace legacy index() with strchr() and avoid calculating the index twice https://bugs.freedesktop.org/show_bug.cgi?id=32276 mcaststreamtransmitter: Fix error message shmtransmitter: Remove unused header includes Update gtk-doc-plugins.mak from common/ Verify the sanity of arguments passed to user-facing functions rtpsession: Unblock pad if the discovery callback is called while disposing of a session docs: Add docs for the shm transmitter docs: Update custom doc building rules to match newer gst tools nice: Use the right enum type for pad link return Version 0.0.22.1
2011-01-13png shlib name changed for png>=1.5.0, so bump PKGREVISIONs.wiz1-1/+2
2010-11-23Update to 0.0.22:wiz1-3/+2
Version 0.0.22 Disable the test for changing the DTMF PT for now python: Require pygobject 2.16 to build rtpconference: The ptime/maxptime in caps are actually uints, not strings Update common and tabify Makefiles gitignore: Hide shm test readme: bump -bad requirement for shm plugin tests: Whitelist shm plugin tests: Clear GError* between tests shmtrans: Don't try to unref NULL pointer on error configure: Require GLib 2.16 for GIO GIO is required by the shm example, require it. tests shm: check that prepared is called shmtrans: Sync downstream element states before linking them shmtrans: Add debug shmtrans: Release teepad before stopping downstream elements shmtrans: Emit local candidate with new path shmstreamtrans: Set the sending in set property (not get) shmtrans: Set do-timestamp and is-live to true on shmsrc shmstreamtransmitter: Emit local-candidates-prepared shm: Document shm stream transmitter shmstream: Also ignore usernames that are empty shm: Replace base_ip with username simplecall: Add shm version of simple-call shm: Verify the success of state changes tests: Add tests for the shm transmitter shm: Implement shm transmitter shm: Add empty transmitter tests: Unlock lock in all cases fsplugin: Release lock on errors elementaddednotifier: Don't abort on elements that have no factory rtpsession: Use copy of codec because mutex has been unlocked Can't use the ca pointer because it is part of a list that has been unlocked. tests: Skip theora reception test if theora is not detected
2010-11-15PKGREVISION bumps for changes to gtk2, librsvg, libbonobo and libgnomeabs1-2/+2
2010-09-14Bump dependency on pixman to 0.18.4 because cairo-1.10 needs thatwiz1-1/+2
version, and bump all depends. Per discussion on pkgsrc-changes.
2010-08-05update to 0.0.21drochner1-7/+2
changes: bugfixes
2010-06-13Bump PKGREVISION for libpng shlib name change.wiz1-2/+2
Also add some patches to remove use of deprecated symbols and fix other problems when looking for or compiling against libpng-1.4.x.
2010-05-05Fix build with libnice-0.0.11 and depend on it.wiz1-2/+3
Bump PKGREVISION. Fixes PR 43241 by Muhammad Hallaj Subery.
2010-02-10Bump revision for PYTHON_VERSION_DEFAULT change.joerg1-2/+2
2010-01-20Bump PKGREVISION for gupnp/gssdp API changes.wiz1-1/+2
2010-01-20Update to 0.0.17:wiz1-2/+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
2009-10-31Update to 0.0.16:wiz1-3/+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
2009-09-23Remove "PYTHON_VERSIONS_ACCEPTED= 26 25 24" which is unnecessarytron1-2/+1
after Python 2.3 has been removed from "pkgsrc". Approved by Thomas Klausner.
2009-08-26bump revision because of graphics/jpeg updatesno1-1/+2
2009-08-17Initial import of farsight2-0.0.14:wiz1-0/+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.
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