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2008-05-26gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the ↵Wim Taymans3-38/+61
jitterbuffer if the gap is too big, we need ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): When checking the seqnum, reset the jitterbuffer if the gap is too big, we need to do this so that we can better handle a restarted source. Fix some comments. * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew), (rtp_jitter_buffer_insert): Tweak the skew resync diff. Use our working seqnum compare function in -base. Rework the jitterbuffer insert code to make it clearer and more performant by only retrieving the seqnum of the input buffer once and by adding some G_LIKELY compiler hints. Improve debugging for duplicate packets. * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp): Fix a comment, we don't do skew correction here..
2008-05-26gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to ↵Håvard Graff1-0/+24
the jitterbuffers when they are changed o... Original commit message from CVS: Patch by: Håvard Graff <havard dot graff at tandberg dot com> * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_propagate_property_to_jitterbuffer), (gst_rtp_bin_set_property): Propagate the do-lost and latency properties to the jitterbuffers when they are changed on rtpbin.
2008-05-26Don't use _gst_pad().Wim Taymans1-3/+3
Original commit message from CVS: * examples/switch/switcher.c: (switch_timer): * gst/replaygain/gstrgvolume.c: (gst_rg_volume_init): * gst/rtpmanager/gstrtpclient.c: (create_stream): * gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp), (gst_sdp_demux_stream_configure_udp_sink): * tests/check/elements/deinterleave.c: (GST_START_TEST), (pad_added_setup_data_check_float32_8ch_cb): * tests/check/elements/rganalysis.c: (send_eos_event), (send_tag_event): Don't use _gst_pad().
2008-05-16docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.Jan Schmidt1-4/+6
Original commit message from CVS: * docs/Makefile.am: Don't attempt to build plugin docs when they're disabled. * gst/bayer/Makefile.am: Add libgstvideo to the link. * gst/rtpmanager/Makefile.am: Fix link order, and move LIBS things to _LIBS
2008-05-14gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a ↵Wim Taymans1-4/+4
warning instead of just posting an error and ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Simply drop bad RTP packets with a warning instead of just posting an error and stopping. This is a perfectly recoverable event and we don't force people to use an rtpbin to filter out bad packets first.
2008-05-13gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.Wim Taymans1-0/+5
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): Actually add the do-lost property to the object.
2008-05-12gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) ↵Wim Taymans1-2/+8
duration when the last packet has a lower ti... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Avoid waiting for a negative (huge) duration when the last packet has a lower timestamp than the current packet.
2008-05-12gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned ↵Peter Kjellerstedt1-0/+3
by gst_pad_get_parent() to prevent a memor... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src): Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memory leak.
2008-05-12gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to ↵Jan Schmidt1-1/+1
avoid compiler warning. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
2008-05-09gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to ↵Peter Kjellerstedt1-0/+2
prevent a memory leak. Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (rtp_source_finalize): Make sure to unref the caps used by RTPSource to prevent a memory leak.
2008-05-08gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our ↵Olivier Crete1-0/+8
callbacks. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/rtpsession.c: (source_clock_rate), (rtp_session_process_bye), (rtp_session_send_bye_locked): Unlock the session lock when calling one of our callbacks. Fixes #532011.
2008-05-08gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.Sjoerd Simons1-0/+1
Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink): Send RTP BYE command on EOS. Fixes bug #531955.
2008-04-25gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too.Wim Taymans2-2/+18
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Expose new jitterbuffer property in rtpbin too.
2008-04-25gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost ↵Wim Taymans1-14/+52
events by default and make a property to ena... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Disable sending out rtp packet lost events by default and make a property to enabe it. We will likely enable it by default when the base depayloaders have a default handler for them so that we don't send these events all through the pipeline for now.
2008-04-25gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function ↵Wim Taymans1-37/+109
that is in -base now. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Remove private version of a function that is in -base now. Add src event handler. Rework the jitterbuffer pushing loop so that it can quickly react to lost packets and instruct the depayloader of them. This can then be used to implement error concealment data.
2008-04-25gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the ↵Wim Taymans1-0/+33
RTCP and sync pads because the defaults ... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink), (create_send_rtcp_src): Set up some internal links functions for the RTCP and sync pads because the defaults are really not correct. Implement a query handler for the RTCP src pad, mostly to correctly report about the latency.
2008-04-25gst/rtpmanager/: Also keep track of the first buffer timestamp together with ↵Wim Taymans5-1/+11
the first Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain): * gst/rtpmanager/rtpsession.c: (update_arrival_stats), (rtp_session_process_sr), (rtp_session_on_timeout): * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Also keep track of the first buffer timestamp together with the first RTP timestamp as they both are needed to construct the timing of outgoing packets in the jitterbuffer and are therefore also needed to manage lip-sync. This fixes lip-sync if the first RTP packets arrive with a wildly different gap.
2008-04-21gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.Olivier Crete4-14/+41
Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (new_ssrc_pad_found): Ref caps when inserting into the cache. Don't leak pads. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_get_clock_rate), (gst_rtp_jitter_buffer_query): Avoid a caps leak. Don't leak refcount in query. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps), (gst_rtp_pt_demux_chain): Avoid caps leaks. * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure), (gst_rtp_session_init), (return_true), (gst_rtp_session_clear_pt_map), (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate): Ref caps when inserting into the cache. Fix some more caps leaks. Fixes #528245.
2008-04-17gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a ↵Wim Taymans4-5/+28
refcount leak. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (get_pt_map), (free_client), (gst_rtp_bin_associate), (gst_rtp_bin_get_free_pad_name): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_get_clock_rate): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate): Unset GValues after g_signal_emitv so that we avoid a refcount leak. Don't leak a padname. Don't leak client streams list. Lock rtpbin when associating streams. Fixes #528245.
2008-04-09gst/rtpmanager/: Avoid leaking pads in the RTP manager.Peter Kjellerstedt2-0/+24
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (free_session): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize): Avoid leaking pads in the RTP manager.
2008-03-11gst/rtpmanager/rtpsession.*: Implement collision and loop detection in ↵Olivier Crete4-22/+253
rtpmanager. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/rtpsession.c: (find_add_conflicting_addresses), (check_collision), (obtain_source), (rtp_session_create_new_ssrc), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye), (rtp_session_send_bye_locked), (rtp_session_send_bye), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Implement collision and loop detection in rtpmanager. Fixes #520626. * gst/rtpmanager/rtpsource.c: (rtp_source_reset), (rtp_source_init): * gst/rtpmanager/rtpsource.h: Add method to reset stats.
2008-03-11gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP ↵Ole André Vadla Ravnås1-4/+36
thread in PAUSED because it might be blocked d... Original commit message from CVS: Based on patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init), (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread), (join_rtcp_thread), (gst_rtp_session_change_state): Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked downstream. Also avoid spawning multiple rtcp threads. Fixes #520894.
2008-03-11gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we ↵Stefan Kost1-1/+1
have no timestamps. Original commit message from CVS: Patch by: Stefan Kost <ensonic@users.sf.net> * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Don't try to reset the clock skew when we have no timestamps. Fixes #519005.
2008-02-20gst/rtpmanager/gstrtpbin.c: Fix small memory leak, leaking caps. Fixes #bug ↵Olivier Crete1-0/+2
517571. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Fix small memory leak, leaking caps. Fixes #bug 517571.
2008-02-14gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet ↵Olivier Crete1-1/+4
when doing synchronisation. Fixes #516160. Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate): Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160.
2008-01-29gst/rtpmanager/gstrtpjitterbuffer.c: Try to get the new clock-rate from the ↵Thijs Vermeir1-0/+7
buffer caps when we receive a new payload... Original commit message from CVS: Patch by: Thijs Vermeir <thijsvermeir at gmail dot com> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Try to get the new clock-rate from the buffer caps when we receive a new payload type instead of always firing the signal. Fixes #512774.
2008-01-25gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not ↵Olivier Crete1-3/+47
provided with caps but with a signal. Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (create_stream), (payload_type_change), (new_ssrc_pad_found): Also handle lip-sync when the clock-rate is not provided with caps but with a signal.
2008-01-25gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and ↵Olivier Crete3-32/+11
extend it so that a clock-rate can be provided... Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain): * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided with each buffer instead. Fixes #511686.
2008-01-25gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.Olivier Crete1-6/+9
Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Remove old unused variable. Track pt on input buffers and get the clock-rate when it changes. Ignore packets with unknown clock-rate. See #511686.
2008-01-25gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. ↵Olivier Crete1-1/+1
Fixes #511920 Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. Fixes #511920
2008-01-11gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to ↵Wim Taymans1-4/+3
parse the clock-rate instead of returning... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate): If we find the caps in the cache, use it to parse the clock-rate instead of returning an error. Fixes a TODO as found by Youness Alaoui.
2008-01-11gst/rtpmanager/: Make it possible to use different user_data for each of the ↵Youness Alaoui3-22/+174
callbacks. Original commit message from CVS: Patch by: Youness Alaoui <youness dot alaoui at collabora dot co dot uk> * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate): * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks), (rtp_session_set_process_rtp_callback), (rtp_session_set_send_rtp_callback), (rtp_session_set_send_rtcp_callback), (rtp_session_set_sync_rtcp_callback), (rtp_session_set_clock_rate_callback), (rtp_session_set_reconsider_callback), (source_push_rtp), (source_clock_rate), (rtp_session_process_bye), (rtp_session_process_rtcp), (rtp_session_send_bye), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Make it possible to use different user_data for each of the callbacks. Fixes #508587.
2008-01-10gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patchThijs Vermeir1-2/+2
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
2008-01-10gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)Thijs Vermeir1-4/+46
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)
2008-01-09gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do ↵Wim Taymans1-1/+0
everything the upsteam peer pad can renegot... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink): Don't set fixed caps, we can basically do everything the upsteam peer pad can renegotiate to. Fixes #507940.
2008-01-04gst/rtpmanager/gstrtpjitterbuffer.c: Don't unref the popped buffer when we ↵Wim Taymans1-3/+1
don't have ownership. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Don't unref the popped buffer when we don't have ownership. Fixes #507020.
2007-12-31gst/rtpmanager/gstrtpssrcdemux.c: Don't clean up pads when going to PAUSED.Wim Taymans1-0/+1
Original commit message from CVS: * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_change_state): Don't clean up pads when going to PAUSED.
2007-12-12gst/rtpmanager/: Clean up the dynamic pads when going to READY.Wim Taymans2-30/+41
Original commit message from CVS: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release), (gst_rtp_pt_demux_change_state): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset), (gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query), (gst_rtp_ssrc_demux_change_state): Clean up the dynamic pads when going to READY.
2007-12-12gst/rtpmanager/: Fix some leaks.Wim Taymans3-0/+17
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_finalize), (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string), (gst_rtp_bin_handle_message): * gst/rtpmanager/rtpsession.c: (rtp_session_finalize), (rtp_session_send_bye): * gst/rtpmanager/rtpsource.c: (rtp_source_finalize): Fix some leaks.
2007-12-10gst/rtpmanager/: Post a message when the SDES infor changes for a source.Wim Taymans4-3/+118
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init), (gst_rtp_bin_handle_message): * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure), (on_ssrc_sdes): Post a message when the SDES infor changes for a source. * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: Update some comments.
2007-12-10gst/rtpmanager/: Add signal to notify of an SDES change.Wim Taymans20-38/+85
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_ssrc_sdes), (create_session), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (on_ssrc_sdes), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_ssrc_sdes), (rtp_session_process_sdes): * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: Add signal to notify of an SDES change. Fix object type in the signal callbacks.
2007-12-10gst/rtpmanager/gstrtpbin.*: Expose SDES items as properties and configure ↵Wim Taymans3-2/+200
the session managers with them. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name), (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Expose SDES items as properties and configure the session managers with them. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_set_property): Fix SSRC property.
2007-12-10gst/rtpmanager/: Update comment.Wim Taymans7-322/+996
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session): * gst/rtpmanager/rtpjitterbuffer.c: Update comment. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property): Define some GObject properties to set SDES and other configuration. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (rtp_session_set_property), (rtp_session_get_property), (on_ssrc_sdes), (rtp_session_set_bandwidth), (rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction), (rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string), (rtp_session_get_sdes_string), (obtain_source), (rtp_session_get_internal_source), (rtp_session_process_sdes), (rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes), (is_rtcp_time): * gst/rtpmanager/rtpsession.h: Add signal when new SDES infor has been found for a source. Create properties for SDES and other info. Simplify the SDES API. Add method for getting the internal source object of the session. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_finalize), (rtp_source_set_property), (rtp_source_get_property), (rtp_source_set_callbacks), (rtp_source_get_ssrc), (rtp_source_set_as_csrc), (rtp_source_is_as_csrc), (rtp_source_is_active), (rtp_source_is_validated), (rtp_source_is_sender), (rtp_source_received_bye), (rtp_source_get_bye_reason), (rtp_source_set_sdes), (rtp_source_set_sdes_string), (rtp_source_get_sdes), (rtp_source_get_sdes_string), (rtp_source_get_new_sr), (rtp_source_get_new_rb): * gst/rtpmanager/rtpsource.h: Add GObject properties for various things. Don't leak the bye reason.
2007-11-22gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer can buffer an unlimited ↵Wim Taymans1-4/+2
amount of time and thus has no max_latency ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_query): jitterbuffer can buffer an unlimited amount of time and thus has no max_latency requirements.
2007-11-02gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).Ole André Vadla Ravnås1-3/+3
Original commit message from CVS: Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> * gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
2007-10-09gst/rtpmanager/gstrtpbin.c: Fix memleak. Fixes #484990.Laurent Glayal1-0/+1
Original commit message from CVS: Patch by: Laurent Glayal <spglegle at yahoo dot fr> * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_class_init): Fix memleak. Fixes #484990.
2007-10-08gst/: Fix compiler warnings shown by Forte.Jan Schmidt1-3/+3
Original commit message from CVS: * gst/librfb/rfbbuffer.c: (rfb_buffer_new_and_alloc): * gst/librfb/rfbbuffer.h: * gst/librfb/rfbdecoder.c: (rfb_socket_get_buffer): * gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_chain): * gst/nsf/nes6502.c: (nes6502_execute): * gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps): * gst/real/gstrealvideodec.c: (open_library): * gst/real/gstrealvideodec.h: * gst/rtpmanager/gstrtpsession.c: (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink): Fix compiler warnings shown by Forte.
2007-10-08gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.Wim Taymans4-46/+90
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (get_pt_map), (gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init): Fix caps refcounting for payload maps. When clearing payload maps, also clear sessions and streams payload maps. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps), (gst_rtp_pt_demux_clear_pt_map), (gst_rtp_pt_demux_chain), (find_pad_for_pt): Implement clearing the payload map. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink): Forward flush events instead of leaking them. * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_rtcp_sink_event): Correctly refcount events before pushing them.
2007-10-05gst/rtpmanager/rtpsession.c: When reconsidering RTCP timeouts, set the next ↵Wim Taymans1-11/+32
timeout against the last report time inst... Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_next_timeout), When reconsidering RTCP timeouts, set the next timeout against the last report time instead of the current clock time so that we don't end up reconsidering forever.
2007-10-05gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead ↵Wim Taymans5-46/+35
of popping it off, which allows us to grea... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Only peek at the tail element instead of popping it off, which allows us to greatly simplify things when the tail element changes. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_recv_rtp_sink): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_sink_event): Forward FLUSH events instead of leaking them. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Remove the tail-changed callback in favour of a simple boolean when we insert a buffer in the queue. Add method to peek the tail of the buffer.
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