Commit graph

3292 commits

Author SHA1 Message Date
Wim Taymans
5c89bb2ab3 gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
(gst_rtp_session_event_send_rtp_sink):
Send EOS when the session object instructs us to.
* gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible for the session manager to instruct us to send EOS. We
currently will EOS when the session is a sender and when the sender part
goes EOS. This is not entirely correct behaviour because the session
could still participate as a receiver.
Fixes #549409.
2009-08-11 02:30:37 +01:00
Wim Taymans
62ecaee748 gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect a gap when the clock_base changed.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
Reset rtp timestamp interpollation when we detect a gap when the
clock_base changed.
Don't try to adjust the ts-offset when it's too big (> 3seconds)
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
* gst/rtpmanager/gstrtpsession.h:
Add method to set session SSRC.
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Added debugging for the collision checks.
Add method to change the internal SSRC of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Reset the clock base when we detect large jumps in the seqnums.
2009-08-11 02:30:37 +01:00
Stefan Kost
cc74738d83 gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Print the pad-name in debug log.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
Use "-" instead of "_" in property names. Can we call them just
"device" like everywhere else?
2009-08-11 02:30:37 +01:00
Olivier Crete
d392defbd3 gst/rtpmanager/gstrtpjitterbuffer.c: Make the buffer metadata writable before inserting it in the jitterbuffer becaus...
Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Make the buffer metadata writable before inserting it in the
jitterbuffer because the jitterbuffer will modify the timestamps.
* gst/rtpmanager/rtpjitterbuffer.c:
Update method comment about requiring writable metadata on buffers.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_rtcp):
Make the RTCP buffer metadata writable because we want to modify the
metadata.
Fixes #546312.
2009-08-11 02:30:37 +01:00
Håvard Graff
1bef5a8ab8 gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Fix debug by logging the right seqnum.
2009-08-11 02:30:37 +01:00
Olivier Crete
2707a84d78 gst/rtpmanager/gstrtpbin.c: Release lock before emitting the request-pt-map signal.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (get_pt_map):
Release lock before emitting the request-pt-map signal.
Fixes #543480.
2009-08-11 02:30:37 +01:00
Peter Kjellerstedt
fd44690d4f gst/rtpmanager/: Corrected a typo (interpollate -> interpolate).
Original commit message from CVS:
* ChangeLog:
* gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr):
Corrected a typo (interpollate -> interpolate).
2009-08-11 02:30:36 +01:00
Peter Kjellerstedt
e2f49d9ccf gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
(gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_send_rtp):
* gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp):
Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
pipeline is running normally.
2009-08-11 02:30:36 +01:00
Peter Kjellerstedt
ca15984e14 gst/rtpmanager/: Do not mix the use of g_get_current_time() with gst_clock_get_time().
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (check_collision),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_send_bye_locked), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_report_blocks), (session_cleanup),
(is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Do not mix the use of g_get_current_time() with gst_clock_get_time().
2009-08-11 02:30:36 +01:00
Stefan Kost
a71ffc55d8 Final round of doc updates.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
2009-08-11 02:30:36 +01:00
Stefan Kost
138c2b7cf9 gst/: More doc updates. More xrefs.
Original commit message from CVS:
* gst/deinterlace/gstdeinterlace.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/sdp/gstsdpdemux.c:
More doc updates. More xrefs.
2009-08-11 02:30:36 +01:00
Stefan Kost
2d1ccbf52e Do not use short_description in section docs for elements. We extract them from element details and there will be war...
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
2009-08-11 02:30:36 +01:00
Wim Taymans
8dc879f15e gst/rtpmanager/gstrtpbin.c: Fix deadlock when shutting down, use a new lock instead to properly shutdown.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_change_state):
Fix deadlock when shutting down, use a new lock instead to properly
shutdown.
2009-08-11 02:30:36 +01:00
Wim Taymans
fda8195d76 gst/rtpmanager/gstrtpbin.c: Break out of callbacks when we are shutting down.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_change_state), (new_payload_found),
(new_ssrc_pad_found):
Break out of callbacks when we are shutting down.
Make sure no state changes can happen when we reconfigure.
2009-08-11 02:30:36 +01:00
Wim Taymans
bd1e0ebfc0 gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the 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..
2009-08-11 02:30:36 +01:00
Håvard Graff
b889dfad30 gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to 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.
2009-08-11 02:30:36 +01:00
Wim Taymans
6716231857 Don't use _gst_pad().
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().
2009-08-11 02:30:35 +01:00
Jan Schmidt
4e5347c8fe docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.
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
2009-08-11 02:30:35 +01:00
Wim Taymans
2506d13ecc gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a 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.
2009-08-11 02:30:35 +01:00
Wim Taymans
cd00eb71b4 gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
Actually add the do-lost property to the object.
2009-08-11 02:30:35 +01:00
Wim Taymans
71c2510665 gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) 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.
2009-08-11 02:30:35 +01:00
Peter Kjellerstedt
fd8061784a gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned 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.
2009-08-11 02:30:35 +01:00
Jan Schmidt
95ab282083 gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to 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.
2009-08-11 02:30:35 +01:00
Peter Kjellerstedt
b1ef03968a gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to 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.
2009-08-11 02:30:35 +01:00
Olivier Crete
bddddbd409 gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our 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.
2009-08-11 02:30:35 +01:00
Sjoerd Simons
c466ae6bdc gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.
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.
2009-08-11 02:30:35 +01:00
Wim Taymans
d6c8809739 gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too.
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.
2009-08-11 02:30:35 +01:00
Wim Taymans
250c38a5ce gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost 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.
2009-08-11 02:30:35 +01:00
Wim Taymans
e2ab966d14 gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function 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.
2009-08-11 02:30:34 +01:00
Wim Taymans
a05b42ef04 gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the 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.
2009-08-11 02:30:34 +01:00
Wim Taymans
e779adca69 gst/rtpmanager/: Also keep track of the first buffer timestamp together with 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.
2009-08-11 02:30:34 +01:00
Olivier Crete
3c5cf0cd38 gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.
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.
2009-08-11 02:30:34 +01:00
Wim Taymans
4cc70a0c22 gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a 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.
2009-08-11 02:30:34 +01:00
Peter Kjellerstedt
959c341cbd gst/rtpmanager/: Avoid leaking pads in the RTP manager.
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.
2009-08-11 02:30:34 +01:00
Olivier Crete
3f58847080 gst/rtpmanager/rtpsession.*: Implement collision and loop detection in 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.
2009-08-11 02:30:34 +01:00
Ole André Vadla Ravnås
6ba2fcd4ff gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP 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.
2009-08-11 02:30:34 +01:00
Stefan Kost
52cdd3c59a gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we 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.
2009-08-11 02:30:34 +01:00
Olivier Crete
db8bdc8b92 gst/rtpmanager/gstrtpbin.c: Fix small memory leak, leaking caps. Fixes #bug 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.
2009-08-11 02:30:34 +01:00
Olivier Crete
a301c9a22b gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet 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.
2009-08-11 02:30:34 +01:00
Thijs Vermeir
b638626053 gst/rtpmanager/gstrtpjitterbuffer.c: Try to get the new clock-rate from the 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.
2009-08-11 02:30:33 +01:00
Olivier Crete
7b2446b676 gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not 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.
2009-08-11 02:30:33 +01:00
Olivier Crete
41ada27f2e gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and 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.
2009-08-11 02:30:33 +01:00
Olivier Crete
eb0993af12 gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.
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.
2009-08-11 02:30:33 +01:00
Olivier Crete
0369f87020 gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. 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
2009-08-11 02:30:33 +01:00
Wim Taymans
6e6c59a198 gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to 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.
2009-08-11 02:30:33 +01:00
Youness Alaoui
03d9faf5fa gst/rtpmanager/: Make it possible to use different user_data for each of the 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.
2009-08-11 02:30:33 +01:00
Thijs Vermeir
c6d892420a gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Fix documentation for latest patch
2009-08-11 02:30:33 +01:00
Thijs Vermeir
a4db9d0943 gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Allow request_new_pad with name NULL (bug #508515)
2009-08-11 02:30:33 +01:00
Wim Taymans
c7818b0c0f gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do 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.
2009-08-11 02:30:33 +01:00
Wim Taymans
c5e9700eda gst/rtpmanager/gstrtpjitterbuffer.c: Don't unref the popped buffer when we 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.
2009-08-11 02:30:33 +01:00
Wim Taymans
cba910a430 gst/rtpmanager/gstrtpssrcdemux.c: Don't clean up pads when going to PAUSED.
Original commit message from CVS:
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_change_state):
Don't clean up pads when going to PAUSED.
2009-08-11 02:30:32 +01:00
Wim Taymans
a965ebff09 gst/rtpmanager/: Clean up the dynamic pads when going to READY.
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.
2009-08-11 02:30:32 +01:00
Wim Taymans
df55cf2f08 gst/rtpmanager/: Fix some leaks.
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.
2009-08-11 02:30:32 +01:00
Wim Taymans
771ed2339d gst/rtpmanager/: Post a message when the SDES infor changes for a source.
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.
2009-08-11 02:30:32 +01:00
Wim Taymans
49e501a647 gst/rtpmanager/: Add signal to notify of an SDES change.
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.
2009-08-11 02:30:32 +01:00
Wim Taymans
95d1f62397 gst/rtpmanager/gstrtpbin.*: Expose SDES items as properties and configure 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.
2009-08-11 02:30:32 +01:00
Wim Taymans
1971ae0d82 gst/rtpmanager/: Update comment.
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.
2009-08-11 02:30:32 +01:00
Wim Taymans
1a8f489093 gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer can buffer an unlimited 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.
2009-08-11 02:30:32 +01:00
Ole André Vadla Ravnås
c5fdb6bff3 gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
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).
2009-08-11 02:30:32 +01:00
Laurent Glayal
8da59edc68 gst/rtpmanager/gstrtpbin.c: Fix memleak. Fixes #484990.
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.
2009-08-11 02:30:31 +01:00
Jan Schmidt
c924d4a466 gst/: Fix compiler warnings shown by Forte.
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.
2009-08-11 02:30:31 +01:00
Wim Taymans
4556ccb666 gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.
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.
2009-08-11 02:30:31 +01:00
Wim Taymans
76a89b5e50 gst/rtpmanager/rtpsession.c: When reconsidering RTCP timeouts, set the next 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.
2009-08-11 02:30:31 +01:00
Wim Taymans
387f41e157 gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead 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.
2009-08-11 02:30:31 +01:00
Wim Taymans
b09507ab0c gst/rtpmanager/gstrtpjitterbuffer.c: Remove some old unused variables.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state), (apply_offset),
(gst_rtp_jitter_buffer_loop):
Remove some old unused variables.
Don't add the latency to the skew corrected timestamp, latency is only
used to sync against the clock.
Improve debugging.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(rtp_jitter_buffer_reset_skew), (calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Handle case where server timestamp goes backwards or wildly jumps by
temporarily pausing the skew correction.
Improve debugging.
2009-08-11 02:30:30 +01:00
Wim Taymans
9c867a2160 gst/rtpmanager/gstrtpbin.c: Fix crasher in dispose.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_client):
Fix crasher in dispose.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Handle cases where input buffers have no timestamps so that no clock
skew can be calculated, in this case interpollate timestamps based on
rtp timestamp and assume a 0 clock skew.
2009-08-11 02:30:30 +01:00
Wim Taymans
2b1f49a26e gst/rtpmanager/gstrtpjitterbuffer.c: Remove jitter correction code, it's now in the lower level object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query):
Remove jitter correction code, it's now in the lower level object.
Use new -core method for doing a peer query.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(calculate_skew), (rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Move jitter correction to the lowlevel jitterbuffer.
Increase the max window size.
When filling the window, already start estimating the skew using a
parabolic weighting factor so that we have a much better startup
behaviour that gets more accurate with the more samples we have.
Increase the default weighting factor for the steady state to get
smoother timestamps.
2009-08-11 02:30:30 +01:00
Wim Taymans
fa00695a39 gst/rtpmanager/gstrtpbin.c: Fix cleanup crasher.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose),
(gst_rtp_bin_finalize):
Fix cleanup crasher.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Dynamically adjust the skew calculation window so that we calculate it
over a period of around 2 seconds.
2009-08-11 02:30:30 +01:00
Wim Taymans
949f1685ce gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. Fixes #478566.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_ssrc_active), (create_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpsession.c: (on_ssrc_active),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_ssrc_active), (rtp_session_process_rb):
* gst/rtpmanager/rtpsession.h:
Add notification of active SSRCs to various RTP elements. Fixes #478566.
2009-08-11 02:30:30 +01:00
Wim Taymans
56d5832287 gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
Link to the right pads regardless of which one was created first in the
ssrc demuxer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsource.c: (calculate_jitter):
Improve debugging.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
(gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links):
* gst/rtpmanager/gstrtpssrcdemux.h:
Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
2009-08-11 02:30:30 +01:00
Wim Taymans
b2aa36cb0d gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
Use lock to protect variable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop):
Reconstruct GST timestamp from RTP timestamps based on measured clock
skew and sync offset.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(rtp_jitter_buffer_set_tail_changed),
(rtp_jitter_buffer_set_clock_rate),
(rtp_jitter_buffer_get_clock_rate), (calculate_skew),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek):
* gst/rtpmanager/rtpjitterbuffer.h:
Measure clock skew.
Add callback to be notfied when a new packet was inserted at the tail.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Remove clock skew detection, it's move to the jitterbuffer now.
2009-08-11 02:30:30 +01:00
Wim Taymans
0441ef80b0 gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
Also set NTP base time on new sessions.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Use the right lock to protect our variables.
Fix some comment.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp),
(gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
Implement getcaps on the sender sinkpad so that payloaders can negotiate
the right SSRC.
2009-08-11 02:30:30 +01:00
Wim Taymans
a93348cc6d gst/rtpmanager/: Various leak fixes.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (free_session),
(get_client), (free_client), (gst_rtp_bin_associate),
(free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose),
(gst_rtp_bin_finalize):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_finalize):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize),
(gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose):
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
* gst/rtpmanager/rtpsession.h:
Various leak fixes.
2009-08-11 02:30:30 +01:00
Wim Taymans
919deb4490 gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so that we can generate better
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base),
(gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp):
Calculate and configure the NTP base time so that we can generate better
NTP times in SR packets.
Set caps on new ghostpad.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Clean debug statement.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Add ntp-ns-base property to convert running_time to NTP time.
Handle NEWSEGMENT events on send and recv RTP pads so that we can
calculate the running time and thus NTP time of the packets.
Simplify getting the current NTP time using the pipeline clock.
Implement internal links functions.
Use the buffer timestamp to calculate the NTP time instead of the clock.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links),
(gst_rtp_ssrc_demux_src_query):
* gst/rtpmanager/gstrtpssrcdemux.h:
Implement internal links function.
Calculate the diff between different streams, this might be used later
to get the inter stream latency.
* gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp):
Simple cleanup.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr):
Make the clock skew window a little bigger.
Apply the clock skew to all buffers, not just one with a new timestamp.
Calculate and debug sender clock drift.
Use extended last timestamp to interpollate for SR reports.
2009-08-11 02:30:29 +01:00
Tim-Philipp Müller
aa8985d1e4 gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562).
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
Make compiler happy: fix compilation with -Wall -Werror
(#473562).
2009-08-11 02:30:29 +01:00
Wim Taymans
e7b6212c51 gst/rtpmanager/: Updated example pipelines in docs.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
(gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
(create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
Updated example pipelines in docs.
Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
Set the default latency correctly.
Add some more points where we can get caps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Add ts-offset property to control timestamping.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
(gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
(gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Various cleanups.
Feed rtpsession manager with NTP time based on pipeline clock when
handling RTP packets and RTCP timeouts.
Perform all RTCP with the system clock.
Set caps on RTCP outgoing buffers.
* gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
(create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain):
* gst/rtpmanager/gstrtpssrcdemux.h:
Also demux RTCP messages.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rb), (rtp_session_process_sr),
(rtp_session_process_rr), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_send_bye),
(session_start_rtcp), (session_report_blocks), (session_cleanup),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Remove the get_time callback, the GStreamer part will feed us with
enough timing information.
Split sync timing and RTCP timing information.
Factor out common RB handling for SR and RR.
Send out SR RTCP packets for lip-sync.
Move SR and RR packet info generation to the source.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp),
(rtp_source_process_sr), (rtp_source_process_rb),
(rtp_source_get_new_sr), (rtp_source_get_new_rb),
(rtp_source_get_last_sr):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Use caps on incomming buffers to get timing information when they are
there.
Calculate clock scew of the receiver compared to the sender and adjust
the rtp timestamps.
Calculate the round trip in sources.
Do SR and RR calculations in the source.
2009-08-11 02:30:29 +01:00
Wim Taymans
f4e6f22315 gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release buffers from the jitterbuffer so that we can h...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop):
Use extended timestamp to release buffers from the jitterbuffer so that
we can handle the rtp wraparound correctly.
2009-08-11 02:30:29 +01:00
Wim Taymans
c576bcec15 gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Improve Comments.
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
(gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
(gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
(create_send_rtp_sink):
Also parse the sink caps for clock-rate instead of only relying on the
result of the signal.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Make sure we fetch the clock rate for payloads we are sending out so
that we can use it for SR reports.
2009-08-11 02:30:29 +01:00
Wim Taymans
325dac0fc2 gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the session manager so that it can generate ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Distribute synchronisation parameters to the session manager so that it
can generate correct SR packets for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time),
(rtp_session_set_timestamp_sync), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Add methods for setting sync parameters.
Set correct RTP time in SR packets using the sync params.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Record last RTP <-> GST timestamp so that we can use them to convert NTP
to RTP timestamps in SR packets.
2009-08-11 02:30:29 +01:00
Wim Taymans
eb86865a62 gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.
2009-08-11 02:30:29 +01:00
Wim Taymans
6835b966ec gst/rtpmanager/gstrtpjitterbuffer.c: When synchronizing buffers, take peer latency into account.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query):
When synchronizing buffers, take peer latency into account.
Don't try to add our latency to invalid peer max latency values.
2009-08-11 02:30:29 +01:00
Tim-Philipp Müller
10d6ba4d61 Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPF...
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.signals:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE
registers a GType that's different than the GstRTPFoo types that
farsight registers (luckily GType names are case sensitive). Should
finally fix #430664.
2009-08-11 02:30:29 +01:00
Wim Taymans
f13ad91c77 gst/rtpmanager/gstrtpjitterbuffer.c: When drop-on-latency is set but we have no latency configured, just push the buf...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_set_property):
When drop-on-latency is set but we have no latency configured, just push
the buffer as fast as possible.
Fix typo in comment.
2009-08-11 02:30:29 +01:00
Wim Taymans
ce70e0f43e gst/rtpmanager/rtpjitterbuffer.*: Fix undefined overflow prone ts_diff handling.
Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c:
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Fix undefined overflow prone ts_diff handling.
2009-08-11 02:30:28 +01:00
Wim Taymans
076da98efb gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix EOS handling.
Convert some DEBUG into WARNINGs.
Pause task when flushing.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
Use system clock for RTCP session management timeouts.
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
Release the session lock when emiting signals.
2009-08-11 02:30:28 +01:00
Stefan Kost
f24c54f4b5 gst/rtpmanager/rtpjitterbuffer.c: Include stdlib.
Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c:
Include stdlib.
2009-08-11 02:30:28 +01:00
Wim Taymans
cdd82f2a95 gst/rtpmanager/: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some...
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c:
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
(rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
(rtp_jitter_buffer_new), (compare_seqnum),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
(rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove complicated async queue and replace with more simple jitterbuffer
code while also fixing some bugs.
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
(create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
(create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
* gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
(on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
(gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
Use new jitterbuffer code.
Expose some new signals in preparation for handling EOS.
2009-08-11 02:30:28 +01:00
Stefan Kost
366a756552 Add stdlib include (free, atoi, exit).
Original commit message from CVS:
* examples/app/appsrc_ex.c:
* examples/switch/switcher.c:
* ext/neon/gstneonhttpsrc.c:
* ext/timidity/gstwildmidi.c:
* ext/x264/gstx264enc.c:
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/spectrum/demo-audiotest.c:
* gst/spectrum/demo-osssrc.c:
* sys/dvb/gstdvbsrc.c:
Add stdlib include (free, atoi, exit).
2009-08-11 02:30:28 +01:00
Jens Granseuer
31571c8cb2 gst/: Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
Original commit message from CVS:
Patch by: Jens Granseuer  <jensgr at gmx net>
* gst/equalizer/gstiirequalizer.c:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/equalizer/gstiirequalizernbands.c:
* gst/rtpmanager/async_jitter_queue.c:
(async_jitter_queue_push_sorted):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
* gst/switch/gstswitch.c: (gst_switch_chain):
Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
Fixes #450185.
2009-08-11 02:30:28 +01:00
Wim Taymans
0c4fe985b6 Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream),
(gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpclient.c: (create_stream),
(gst_rtp_client_request_new_pad):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpssrcdemux.c:
Rename elements to avoid conflict with farsight elements with the same
name. Fixes #430664.
2009-08-11 02:30:28 +01:00
Wim Taymans
2a8cfc6410 Document stuff.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_clear_pt_map):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_clear_pt_map):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Document stuff.
Add clear-pt-map action signal where needed.
2009-08-11 02:30:27 +01:00
Wim Taymans
3bc059707d gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
We always use fixed caps.
2009-08-11 02:30:27 +01:00
David Schleef
720dfeb3a5 gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. Work around.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
g_hash_table_remove_all() only exists in 2.12.  Work around.
2009-08-11 02:30:27 +01:00
Wim Taymans
62d401eb93 gst/rtpmanager/async_jitter_queue.c: Fix leak when flushing.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c:
(async_jitter_queue_set_flushing_unlocked):
Fix leak when flushing.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add clear-pt-map signal.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop):
Init clock-rate to -1 to mark unknow clock rate.
Fix flushing.
2009-08-11 02:30:27 +01:00
Stefan Kost
15b54ec7e2 gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
Original commit message from CVS:
* gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
qtdemux_parse_segments, qtdemux_parse_trak):
* gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
rtp_session_get_location, rtp_session_get_tool,
rtp_session_process_bye, session_report_blocks):
* gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
* gst/switch/Makefile.am:
Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
2009-08-11 02:30:27 +01:00
Stefan Kost
091c2cfbc0 gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a...
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
async_jitter_queue_set_low_threshold,
async_jitter_queue_length_ts_units_unlocked,
async_jitter_queue_unref_and_unlock, async_jitter_queue_unref,
async_jitter_queue_lock, async_jitter_queue_push,
async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted,
async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop,
async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
async_jitter_queue_set_flushing_unlocked,
async_jitter_queue_unset_flushing_unlocked):
Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
2009-08-11 02:30:27 +01:00
Wim Taymans
88f2441722 gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Pass queries upstream.
2009-08-11 02:30:27 +01:00
Wim Taymans
a241c62ecb gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Add some debug info.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_send_rtp):
Store real user name in the session.
2009-08-11 02:30:27 +01:00
Wim Taymans
600afaaff9 gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
2009-08-11 02:30:26 +01:00
Wim Taymans
e6537bcd7c gst/rtpmanager/gstrtpsession.c: Remove debug.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
Remove debug.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_sdes), (calculate_rtcp_interval),
(rtp_session_next_timeout), (session_report_blocks):
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
Improve debugging
Fix interval for BYE/RTCP packets.
2009-08-11 02:30:26 +01:00
Wim Taymans
a7b80281d1 gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
Move reconsideration code to the rtpsession object.
Simplify timout handling and add reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
(obtain_source), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_bye), (rtp_session_process_rtcp),
(calculate_rtcp_interval), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_start_rtcp),
(session_report_blocks), (session_cleanup), (session_sdes),
(session_bye), (is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Handle timeout of inactive sources and senders.
Implement BYE scheduling.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_process_sr), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add members to check for timeouts.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
(rtp_stats_calculate_bye_interval):
* gst/rtpmanager/rtpstats.h:
Use RFC algorithm for calculating the reporting interval.
2009-08-11 02:30:26 +01:00
Wim Taymans
43f0b878c9 gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Implement forward and reverse reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources),
(rtp_session_get_num_active_sources), (rtp_session_process_sr),
(session_report_blocks):
* gst/rtpmanager/rtpsession.h:
Small cleanups.
2009-08-11 02:30:26 +01:00
Wim Taymans
333764307d gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable.
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst/rtpmanager/gstrtpbin.c: (create_stream),
(gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Make default jitterbuffer latency configurable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Debuging cleanups.
2009-08-11 02:30:26 +01:00
Wim Taymans
ae536e0c89 gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.
2009-08-11 02:30:25 +01:00
Wim Taymans
23883be047 gst/rtpmanager/gstrtpbin.c: fix for pad name change
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.
2009-08-11 02:30:25 +01:00
Tim-Philipp Müller
677b361dc3 gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
2009-08-11 02:30:25 +01:00
Wim Taymans
54b3dec1f5 configure.ac: Disable rtpmanager for now because it depends on CVS -base.
Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
2009-08-11 02:30:25 +01:00
Wim Taymans
490113d40d gst/rtpmanager/: Protect lists and structures with locks.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found),
(create_recv_rtp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_finalize),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_request_new_pad):
Protect lists and structures with locks.
Return FLOW_OK from RTCP messages for now.
2009-08-11 02:30:25 +01:00
Wim Taymans
8bbea77a41 gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (gst_rtp_bin_class_init), (pt_map_requested):
Emit pt map requests and cache results.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps),
(gst_jitter_buffer_sink_setcaps),
(gst_rtp_jitter_buffer_get_clock_rate),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Emit request-pt-map signals.
2009-08-11 02:30:25 +01:00
Wim Taymans
03bf43d50e gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
Some more custom marshallers.
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(clock_rate_request), (create_stream), (gst_rtp_bin_class_init),
(pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp):
* gst/rtpmanager/gstrtpbin.h:
Prepare for caching pt maps.
Connect to signals to collect pt maps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add request_clock_rate signal.
Use scale insteat of scale_int because the later does not deal with
negative numbers.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_chain):
* gst/rtpmanager/gstrtpptdemux.h:
Implement request-pt-map signal.
2009-08-11 02:30:24 +01:00
Wim Taymans
8c67b5d7dd gst/rtpmanager/: Added custom marshallers for signals.
Original commit message from CVS:
* gst/rtpmanager/.cvsignore:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpbin-marshal.list:
Added custom marshallers for signals.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Prepare for emiting pt map signals.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Fix signals.
2009-08-11 02:30:24 +01:00
Wim Taymans
a6aa41dc21 gst/rtpmanager/gstrtpbin.*: Provide a clock.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_init), (gst_rtp_bin_provide_clock):
* gst/rtpmanager/gstrtpbin.h:
Provide a clock.
2009-08-11 02:30:24 +01:00
Wim Taymans
1b0ae2608f gst/rtpmanager/gstrtpbin.c: Fix pad template name parsing.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
Fix pad template name parsing.
2009-08-11 02:30:24 +01:00
Wim Taymans
63dbc75734 gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Add some debug and comments.
Fix double unref() in error cases.
2009-08-11 02:30:24 +01:00
Wim Taymans
9bfc641f0d gst/rtpmanager/gstrtpbin.*: Add debugging category.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (find_stream_by_ssrc), (create_stream),
(gst_rtp_bin_class_init), (new_payload_found),
(new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp):
* gst/rtpmanager/gstrtpbin.h:
Add debugging category.
Added RTPStream to manage stream per SSRC, each with its own
jitterbuffer and ptdemux.
Added SSRCDemux.
Connect to various SSRC and PT signals and create ghostpads, link stuff.
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
Added rtpbin to elements.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix caps and forward GstFlowReturn
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad):
Add debug category.
Add event handling
* gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
(create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Add debug category.
Add new-pt-pad signal.
2009-08-11 02:30:24 +01:00
Wim Taymans
a9d14ed310 gst/rtpmanager/: Added simple SSRC demuxer.
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc),
(create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init),
(gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Added simple SSRC demuxer.
2009-08-11 02:30:23 +01:00
Wim Taymans
5351f0cb51 gst/rtpmanager/: Some more ghostpad magic.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (gst_rtp_bin_base_init), (create_recv_rtp),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
Some more ghostpad magic.
2009-08-11 02:30:23 +01:00
Wim Taymans
fdae491de7 gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly.
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
Add .h file so it can be disted properly.
2009-08-11 02:30:23 +01:00
Wim Taymans
f0d1ab1c1f Add RTP session management elements. Still in progress.
Original commit message from CVS:
* configure.ac:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
(signal_waiting_threads), (async_jitter_queue_ref),
(async_jitter_queue_ref_unlocked),
(async_jitter_queue_set_low_threshold),
(async_jitter_queue_set_high_threshold),
(async_jitter_queue_set_max_queue_length),
(async_jitter_queue_get_g_queue), (calculate_ts_diff),
(async_jitter_queue_length_ts_units_unlocked),
(async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
(async_jitter_queue_lock), (async_jitter_queue_unlock),
(async_jitter_queue_push), (async_jitter_queue_push_unlocked),
(async_jitter_queue_push_sorted),
(async_jitter_queue_push_sorted_unlocked),
(async_jitter_queue_insert_after_unlocked),
(async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
(async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
(async_jitter_queue_length_unlocked),
(async_jitter_queue_set_flushing_unlocked),
(async_jitter_queue_unset_flushing_unlocked),
(async_jitter_queue_set_blocking_unlocked):
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(gst_rtp_bin_class_init), (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
(gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
(free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
(gst_rtp_client_class_init), (gst_rtp_client_init),
(gst_rtp_client_finalize), (gst_rtp_client_set_property),
(gst_rtp_client_get_property), (gst_rtp_client_change_state),
(gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
(gst_jitter_buffer_sink_setcaps), (free_func),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_activate_push),
(gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
(compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
(gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
(gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
(gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (gst_rtp_session_change_state),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
* gst/rtpmanager/gstrtpsession.h:
Add RTP session management elements. Still in progress.
2009-08-11 02:30:23 +01:00
Mark Nauwelaerts
96e72522fc avidemux: push mode; cater for chunk padding 2009-08-10 14:41:52 +02:00
Mark Nauwelaerts
f67db2a089 avidemux: only use stream's pad after having checked it exists 2009-08-10 14:41:34 +02:00
Mark Nauwelaerts
4249f52c6c avidemux: sprinkle some more GST_DEBUG_FUNCPTR 2009-08-10 14:41:29 +02:00
Mark Nauwelaerts
6d26594eef avidemux: post error message if no pads to push EOS event on 2009-08-10 14:41:27 +02:00
Mark Nauwelaerts
b0a0c06155 avidemux: fix typo in warning message 2009-08-10 14:41:23 +02:00
Mark Nauwelaerts
7750173244 avidemux: fix some buffer ref handling 2009-08-10 14:41:19 +02:00
Mark Nauwelaerts
5b0f7f04e7 avidemux: do not exceed maximum number of supported streams 2009-08-10 14:41:16 +02:00
Mark Nauwelaerts
effa7b4660 avidemux: prevent double unref; gst_avi_demux_parse_avih already unrefs 2009-08-10 14:41:14 +02:00
Mark Nauwelaerts
42bc085d95 avidemux: verify size of INFO LIST to satisfy subsequent expectations 2009-08-10 14:41:12 +02:00
Mark Nauwelaerts
f4f8e8532c avidemux: check video stream framerate against avi header frame duration
The former might be bogus in silly cases, and the latter seems to
carry more weight.
2009-08-10 14:41:09 +02:00
Mark Nauwelaerts
3863871100 avidemux: streamline stream duration calculation 2009-08-10 14:41:07 +02:00
Edward Hervey
d29ba8d48f matroska: remove dead assignments 2009-08-10 09:58:33 +02:00
Edward Hervey
0d6f0801f5 rtp: Remove dead assignments and resulting unneeded variables. 2009-08-10 09:58:33 +02:00
Thiago Santos
08862850a7 matroska: Adds support to muxing/demuxing WMA
Adds support for muxing wma audio family and fixes
demuxing of wma family in matroskademux. matroskademux
was broken because it missed codec_data.
2009-08-09 20:34:05 -03:00
Thiago Santos
df442b4727 matroskamux: adds support for wmv family
Adds support to WMV1, WMV2, WMV3 and other family formats that
are signaled by the 'format' field in the caps (i.e. WVC1).
Partially fixes #576378
2009-08-09 20:34:04 -03:00
LoneStar
494555cecd id3demux: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8
Fixes bug #499242.
2009-08-09 12:52:17 +02:00
Vincent Penquerc'h
19b7001bf9 matroska: add kate subtitle support to matroska muxer and demuxer
See #525743.
2009-08-08 12:54:48 +01:00
Tim-Philipp Müller
b0bcb27517 id3demux: add ID3 v2.3 spec as well 2009-08-07 16:51:45 +01:00
Tim-Philipp Müller
0417283077 id3demux: sizes in ID3 v2.3 are unlikely to be sync-safe integers
In ID3 v2.3 compressed frames will have a 4-byte data length indicator
after the frame header to indicate the size of the decompressed data.
This integer is unlikely to be a sync-safe integer for v2.3 tags,
only in v2.4 it's sync-safe.
2009-08-07 16:42:39 +01:00
Tim-Philipp Müller
3ec0a31d64 id3demux: fix typo in debug message 2009-08-07 16:36:55 +01:00
Tim-Philipp Müller
2e05af3876 id3demux: fix parsing of unsync'ed ID3 v2.4 tags and frames
Reversing the unsynchronisation seems to work slightly differently
for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame
sizes in the frame header, so the unsynchronisation is applied to
the whole frame data including all the frame headers. v2.4 frames
have sync-safe sizes, however, so the unsynchronisation only needs
to be applied to the actual frame data, and it seems that's what's
being done as well. So we need to undo the unsynchronisation on a
per-frame basis for v2.4 tags for things to work properly.

Fixes extraction of coverart/images from APIC frames in ID3 v2.4
tags (#588148).

Add unit test for this as well.
2009-08-07 16:02:23 +01:00
Wim Taymans
ddfa9961c6 rtph264pay: use array instead of queue 2009-08-06 11:14:44 +02:00
Mark Nauwelaerts
2bfb42c5f8 rtph264pay: push NALs only after SPS/PPS
parse complete (bytestream) buffer for SPS/PPS before pushing NALs.

Fixes #564501.
2009-08-06 11:14:44 +02:00
Edward Hervey
d65d542e9d rtpqdm2depay: Fix debug statement. 2009-08-04 11:17:17 +02:00
Edward Hervey
20c7977b9b rtpqdm2depay,rtpsv3vdepay: Add debugging category. 2009-08-03 21:26:31 +02:00
Edward Hervey
25c5514fab rtpqdm2depay: Handle gaps in incoming packets.
Whenever we see a gap, we flush the temporary packets (but not the adapter). If we
had some data temporarily stored it will be outputted (the sound will sound a bit
garbled... but that's how it sounds on MacOSX :)
2009-08-03 21:26:30 +02:00
Edward Hervey
6aff520a24 rtpqdmdepay: Fix CRC calculation and remove commented code. 2009-08-03 21:26:30 +02:00
Edward Hervey
d39c057e42 rtp: New QDM2 rtp depayloader.
Reverse-engineered by comparing:
* A rtp hinted file provided by DarwinStreamingServer
* The output procued by DSS for that same file

Also used various streaming sources available on the internet to fine-tune
the code.

The header/codec_data extraction methods are from FFMpeg (LGPL).
2009-08-03 21:26:30 +02:00
Edward Hervey
e2b3665ae6 rtpsv3vdepay: Properly fill codec_data and cleanup code a bite more. 2009-08-03 21:26:30 +02:00
Edward Hervey
65a2871e90 rtpsv3vdepay: Only output buffers once we're configured. 2009-08-03 21:26:30 +02:00
Edward Hervey
1743763c0b rtpsv3vdepay: Add more encoding-name variants 2009-08-03 21:26:30 +02:00
Sebastian Dröge
8b9d547c14 flvmux: Fix writing of the index for < 128 buffers
Partially fixes bug #590447.
2009-08-03 20:08:00 +02:00
Sebastian Dröge
cb4eb5714c flvmux: Fix resetting of the element
Reset the have_video/have_audio flags and make sure to
properly release the request pads.

Partially fixes bug #590447.
2009-08-03 20:07:00 +02:00
Wim Taymans
784b95ddbf rtspsrc: don't add non-utf8 chars to structures 2009-08-03 18:13:46 +02:00
Luc Deschenaux
654ca56d85 jpegdepay: use attributes for extra properties
Use some of the SDP attributes when they are present to specify the output
dimension and framerate. This allows us to receive jpeg frames larger than
2040 width/height.

Fixes #564437
2009-08-03 18:02:31 +02:00
Wim Taymans
efb9c17975 RTP docs: update with attributes in caps 2009-08-03 18:01:27 +02:00
Luc Deschenaux
f96e900a64 rtspsrc: put all SDP attributes on caps
Put the SDP attributes on the caps too so that they can be used by
depayloaders.

See #564437
2009-08-03 17:21:44 +02:00
Tim-Philipp Müller
6c323f5b0d multiudpsink: don't do things with side-effects inside g_return_val_if_fail()
Someone might compile this code with -DG_DISABLE_ASSERT some day.
2009-08-02 11:50:43 +01:00
Tim-Philipp Müller
93690bfdd6 flvmux: fix invalid write caused by using sizeof("string") as length
sizeof("foo") includes the string's NUL-terminator in the size returned,
but we're writing strings here with an explicit size at the beginning
and no NUL-terminator. In most cases using sizeof("foo") as length in
memcpy is not harmful, but it is where the string goes right at the
end of our buffer to write, since we don't allocate space for that
NUL terminator.
2009-07-31 23:54:47 +01:00
Sebastian Dröge
22d712786c avidemux: Fix last commit and improve readability 2009-07-29 14:31:48 +02:00
Руслан Ижбулатов
3702fcdb80 Fixed the fix for TIME->DEFAULT conversion.
Fixes bug #578052 again.
2009-07-29 13:58:33 +02:00
Edward Hervey
050e91995e rtpsv3depay: Fix width/height calculation, bring up to marginal rank.
Based on documentation found on http://wiki.multimedia.cx/
2009-07-29 13:39:08 +02:00
Thiago Santos
52482a3741 avimux: adds support to wma 2009-07-28 00:30:43 -03:00
Thiago Santos
f43b442cf9 avimux: adds support to wmv 2009-07-28 00:07:15 -03:00
Thiago Santos
40abf68562 qtdemux: Downgrade warning message to debug 2009-07-27 21:39:57 -03:00
Sebastian Dröge
f0054bcc82 effectv: Don't allow caps changes for some effectv filters
These filters use information from previous frames to
generate the current frame and a caps change will make
the effect start from the beginning again.
2009-07-24 19:54:05 +02:00
Sebastian Dröge
6eada080a0 warptv: Make the sine table global instead of having it in every instance 2009-07-24 19:54:05 +02:00
Sebastian Dröge
aa02444768 flvdemux: Implement SEEKING query
Also add some more query types to the answer of the query type function.

Fixes bug #589424.
2009-07-23 11:51:07 +02:00
Stefan Kost
8990398733 interleave: fix indenting and upgrade two debugs to warnings.
Fix newlines in variable decls. Change two debugs to become warnings as they
indicate that things will not work.
2009-07-21 10:07:00 +03:00
Sebastian Dröge
b7bf2f6820 matroskademux: Answer SEEKING queries in the original format 2009-07-21 07:52:00 +02:00
Josep Torra
efcfb89b5c udputils: initialize struct content with 0.
Fixes some random crashes.
2009-07-21 01:12:44 +02:00
Sebastian Dröge
bb03d8ff18 matroskademux: Implement SEEKING query 2009-07-20 16:52:19 +02:00
Sebastian Dröge
c5b420068a effectv: Chain up finalize to the parent class in warptv
Fixes a memory leak.
2009-07-16 17:10:21 +02:00
Sebastian Dröge
2abd58de9d effectv: Add rippletv element
This produces a water ripple effect on the video input,
based on motion or a rain drop algorithm.

Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.

Fixes bug #588695.
2009-07-16 12:05:31 +02:00
Sebastian Dröge
433255304e effectv: Add streaktv effect filter element
This combines the StreakTV and BaltanTV filters from the
effectv project.

Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.

Fixes bug #588368.
2009-07-16 12:04:38 +02:00
Sebastian Dröge
f981bec99d effectv: Fix processing on big endian architectures 2009-07-16 12:04:36 +02:00
Sebastian Dröge
c17134c6de effectv: Add radioactv effect filter
This filter adds a radiation-like motion blur effect
to the video stream.

Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.

Fixes bug #588359.
2009-07-16 12:04:08 +02:00
Sebastian Dröge
3ad603be84 effectv: Make the optv threshold property an uint 2009-07-16 12:04:06 +02:00
Sebastian Dröge
2c2611b6bf effect: Add optv effect filter from the effectv project
This filter binarizes input frames and combines them with various
optical pattern.

Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.

Fixes bug #588349.
2009-07-16 12:03:29 +02:00
Marc Leeman
7484b631b7 mpvpay: Rework the timestamping
Rework the timestamping in the mpv payloader so that the timestamps are more
accurate.

Fixes #587680
2009-07-13 17:55:25 +02:00
Sebastian Dröge
91ad86c0f9 videomixer: Random cleanup 2009-07-10 19:54:25 +02:00
Sebastian Dröge
f19ef7eada videomixer: Send queries to the master pad by default instead of all pads 2009-07-10 19:54:13 +02:00
Sebastian Dröge
0bf61ecfaf videomixer: Add RGB, BGR, xRGB, RGBx, xBGR, BGRx support 2009-07-10 19:35:49 +02:00
Sebastian Dröge
bbcb4f8f15 videomixer: Clean up debugging a bit 2009-07-10 17:43:07 +02:00
Sebastian Dröge
0775db4455 videomixer: Remove some redundant checks and error out immediately if not negotiated
Also stop leaking the output buffer in some error cases.
2009-07-10 17:33:40 +02:00
Sebastian Dröge
4ccd9c92ae videomixer: Remove the calculate_frame_size() function and use libgstvideo instead 2009-07-10 17:23:03 +02:00
Edward Hervey
34c97c0c6f videomixer: Remove unused link/unlink pad methods 2009-07-10 14:37:16 +02:00
Edward Hervey
b02949faeb videomixer: I420 mode: Add fast path for 0.0 and 1.0 alpha
If the source alpha is 0.0, we take nothing.
If the source alpha is 1.0, we overwrite everything.
2009-07-10 14:37:13 +02:00
Edward Hervey
3c88249d48 videomixer: I420 blending : Fix main algorithm.
When blending a source layer with an alpha of 'a' on top of another
destination layer we take the sum of:
* 'a' percent of the source layer
* (100 - 'a') percent of the destination layer (the remainder)
2009-07-10 14:37:10 +02:00
Edward Hervey
ace4cb2295 videomixer: Make debugging category global to all the code. 2009-07-10 14:37:07 +02:00
Edward Hervey
3ebf5e9a2a videomixer: improve readability of debugging statements. 2009-07-10 14:37:04 +02:00
Mark Nauwelaerts
a905ef233e rtspsrc: do not leak timeout message 2009-07-09 11:34:40 +02:00
Sebastian Dröge
63115fe72c avi: Don't forward NEWSEGMENT events from upstream
New ones are generated later and simply forwarding them can
result in NEWSEGMENT events of different format going downstream.

Fixes bug #587983.
2009-07-09 07:14:23 +02:00
Sebastian Dröge
356972740a videomixer: Make checker pattern lookup table constant 2009-07-08 18:19:45 +02:00
Sebastian Dröge
69f9b7c8d6 videomixer: Add support for ARGB
And clean up the caps parsing.
2009-07-08 18:17:48 +02:00
Benjamin Gaignard
abd383a2a6 udp: Initialize pointer to NULL
Otherwise we're calling free() with some random
memory address in error cases.

Fixes bug #587982.
2009-07-08 15:19:03 +02:00
Mark Nauwelaerts
977796fd07 qtdemux: sprinkle some more const 2009-07-08 11:20:30 +02:00
Mark Nauwelaerts
a4d586daac qtdemux: perform some more (careful) data buffering
Once buffering has started (with an mdat atom), continue buffering
until moov atom is reached, which handles cases with multiple
mdat atoms.  Also keep adapter/offset better in sync with upstream
and fix some debug statements.  Fixes #587426.
2009-07-08 11:20:27 +02:00
Philip Jgenstedt
0ebff2d14c avidemux: Replace deprecated GST_DISABLE_DEBUG with correct macro. Fixes #587826 2009-07-06 10:40:31 +02:00
Tim-Philipp Müller
2bcf52dde7 qtdemux: error out instead of dividing by 0
Error out if timescale is 0.
2009-07-01 13:07:48 +01:00
Tim-Philipp Müller
f6a1211495 Revert "qtdemux: Make sure we don't blacklist streams by wrongly comparing their"
This reverts commit 5503a59a57.

Reverting this since it causes regressions with a lot of sample files
I have, all of which worked fine with the last -good release (#586891).
2009-07-01 09:32:42 +01:00
Tim-Philipp Müller
ae27524be0 qtdemux: comment out unused structure 2009-07-01 09:24:38 +01:00
Tim-Philipp Müller
8fa148d2f1 qtdemux: more size checks, and use g_try_new0() instead of g_new0()
Whenever we alloc something based on a user-supplied size, we should
really use g_try_new(), otherwise we can easily be made to abort by
passing a ridiculously large number to us for allocing. Fixes
problems with some fuzzed files.
2009-07-01 09:24:38 +01:00
Tim-Philipp Müller
405aae4568 qtdemux: guard against bogus atom sizes and short reads
Check the possibly 64-bit atom size more carefully before casting it
to an int and passing it to gst_pad_pull_range(), otherwise we might
end up pulling 0 bytes, getting an empty buffer as requested and
dereferencing not available data whilst thinking we actually asked
for and got 0x1000000000000 bytes. Similar fix for push mode operation
where neededbytes ends up being 0 bytes, which makes us assert. Fixes
crash with broken or fuzzed file (NB #122378).
2009-07-01 09:24:38 +01:00
Tim-Philipp Müller
c730912f67 qtdemux: use 0x prefix when logging numbers in hex 2009-07-01 09:24:38 +01:00
LRN
122d40a742 Don't use sendmsg()-dependent code on Windows
Fixes #585842
2009-06-30 20:38:33 +02:00
Wim Taymans
89f0c37c9f law: fix caps and negotiation
Fix the caps to include the depth (instead of width twice) in the caps of
audio/x-raw-int.
Fix negotiation to not only copy the rate/channels of the first structure.
2009-06-30 15:59:20 +02:00
Tim-Philipp Müller
1fb30a154a qtdemux: don't process track_num/track_count tags with a 0 value
Number/count values of 0 mean they're not set. Don't put those in the
taglist.
2009-06-26 13:29:27 +01:00
Julien Moutte
5503a59a57 qtdemux: Make sure we don't blacklist streams by wrongly comparing their
duration with entire clip duration.
2009-06-25 13:23:40 +02:00
Krzysztof Błaszkowski
9fbdfefc56 rtpdec: fix some buffer leaks 2009-06-25 13:18:14 +02:00
Edward Hervey
32a3d6e717 flvparse: Add missing break in switch/case. 2009-06-25 08:11:09 +02:00
Edward Hervey
67ca4c57b1 flvdemux: Remove unused variable, hint branch likeliness, add comments. 2009-06-25 08:10:38 +02:00
Edward Hervey
ff3730fb7b avidemux: Removed unused variable 2009-06-25 08:09:57 +02:00
Edward Hervey
4780d17894 qtdemux: Remove dead assignments and unused variables.
Also add branch likeliness macros.
2009-06-25 07:41:44 +02:00
Edward Hervey
b9d7f2527e qtdemux: Fix uninitialized variables. Fixes build on macosx 2009-06-25 07:41:43 +02:00
Tim-Philipp Müller
632bb7818a avidemux: short-circuit gst_avi_demux_src_convert() when parsing the index
Don't call gst_avi_demux_src_convert() for each single index entry. Not
only do we already have the pointer to the stream context, we also know
the formats we want to convert from and to already, so we may just as
well use optimised conversion routines that bypass some of the checks
and lookups made in gst_avi_demux_src_convert().
2009-06-24 13:04:01 +01:00
Edward Hervey
86c2299ed1 qtdemux: Another round of G_*LIKELY micro-optimisations. 2009-06-24 12:48:32 +02:00
Edward Hervey
30dd458567 qtdemux: Take last sample duration for dummy segment calculation.
This fixes the cases where files without EDL wouldn't output their
last buffer.
2009-06-24 12:48:32 +02:00
Edward Hervey
4e6808bc52 avidemux: Sprinkle branch likeliness macros over the code. 2009-06-24 12:37:39 +02:00
Edward Hervey
279be94321 qtdemux: Add GST_MEMDUMP statements for unknown atoms.
This is to help developers track down and implement unhandled atoms faster.
2009-06-24 12:37:38 +02:00
Sebastian Dröge
810c60a6f3 deinterlace: Remove the interlaced field from the output caps if deinterlacing is enabled 2009-06-23 17:52:29 +02:00
Sebastian Dröge
20668a0782 deinterlace: Copy the correct line from correct place in the history 2009-06-23 17:52:29 +02:00
Wim Taymans
81d7a297f7 rtspsrc: use same protocols after redirect
After a redirect we want to use the same protocols that we were using for the
current url.
2009-06-23 16:39:36 +02:00
Tim-Philipp Müller
da4c1c9227 qtdemux: don't leak cover art 2009-06-23 15:35:37 +01:00
Tim-Philipp Müller
0d9dccee4f udp: fix compiler warning about EAI_ADDRFAMILY getting redefined in some cases
Include the header from where we include all the system headers with the
socket stuff before we try to define EAI_ADDRFAMILY ourselves, otherwise
we define it ourselves and then get a compiler warning if a system header
defines it as well without guarding against it being defined already.
2009-06-23 14:10:10 +01:00
Wim Taymans
dff3f37bdf matroska: and the new headers too 2009-06-23 14:39:56 +02:00
Wim Taymans
8a4dc37544 matroske: fix compiler error
change gpointer to guint8 * for codec_state and codec_priv as some
functions operate on those types and it avoids breaking strict-aliasing
rules.
2009-06-23 14:32:43 +02:00
Wim Taymans
9600c54938 matroskademux: avoid leaking buffers
Don't leak buffers when resyncing to a keyframe.
Avoid leaking buffers when exiting the loop on error conditions.
Add some more debug info.

Fixes #585911
2009-06-23 12:42:33 +02:00
Tim-Philipp Müller
ace09d40bd qtdemux: use GST_MEMDUMP 2009-06-22 10:49:03 +01:00
Tim-Philipp Müller
0a8f254460 apedemux: add container-format tag
Use pbutils here because the string is translated.
2009-06-22 10:49:03 +01:00
Tim-Philipp Müller
98fa27dcd1 id3demux: add container-format tag
Using pbutils here because the string is translated.
2009-06-22 10:49:02 +01:00
Tim-Philipp Müller
6c6e96becd multipartdemux: post container-format tag 2009-06-22 10:49:02 +01:00
Tim-Philipp Müller
323517f527 matroska-demux: post container-format tags 2009-06-22 10:49:01 +01:00
Tim-Philipp Müller
4fe23fbe4b avidemux: post container-format tag 2009-06-22 10:49:01 +01:00
Tim-Philipp Müller
b8176ee9cc qtdemux: post container-format tags 2009-06-22 10:49:01 +01:00
Sebastian Dröge
a3cb8f005b audioamplify: Fix integer overflows on 32 bit architectures 2009-06-21 17:13:43 +02:00
Kipp Cannon
f80b62c3db audioamplify: Don't declare a loop index static
The previous patch to add support for additional sample formats possibly
introduced a reentrancy bug:  a variable used for a loop index was declared
static.  This patch fixes that, and also adds a "/* *INDENT-ON* */" annotation
following the macro block.  (I don't know what the annotation is for, but the
adder, where I copied this from, has it).
2009-06-21 09:50:54 +02:00
Sebastian Dröge
ffe64fb934 audioamplify: Fix off-by-one in wrap-positive mode 2009-06-19 22:37:27 +02:00
Kipp Cannon
afccf53ace audioamplify: Add noclip method and support for more formats
Fixes bug #585828 and #585831.
2009-06-19 22:20:45 +02:00
Koop Mast
8ecd5f1a06 udp: Fix build on FreeBSD
Fixes bug #586397.
2009-06-19 21:47:29 +02:00
Ognyan Tonchev
83a0e7d2a3 rtpmp4vpay: add support for buffer-list
See #585559
2009-06-19 18:00:35 +02:00
Ognyan Tonchev
d5d0364d7b rtpjpegpay: add support for buffer-lists
See #585559
2009-06-19 17:57:12 +02:00
Ognyan Tonchev
40ec22788f rtph264pay: add support for buffer-lists
See #585559
2009-06-19 17:53:32 +02:00
Wim Taymans
8f43709e00 udputils: don't free invalid memory
As spotted by benjiG in IRC.
don't free invalid memory when getaddrinfo failed.
2009-06-19 16:00:51 +02:00
Mark Nauwelaerts
3d8f31843c avidemux: streaming; adjust sizes to cater for padding in chunks 2009-06-18 16:59:26 +02:00
Mark Nauwelaerts
08c9019566 avidemux: streaming mode; handle data chunks grouped in rec lists.
Fixes #567983.
2009-06-17 12:31:42 +02:00
Mark Nauwelaerts
71eb4a4dfa qtdemux: map some tags to COMPOSER rather than ARTIST 2009-06-17 12:31:40 +02:00
Mark Nauwelaerts
0d2db95453 qtdemux: fix some 3GP tag extraction (keywords, genre, location) 2009-06-17 12:31:37 +02:00
Mark Nauwelaerts
0f0117aabb qtdemux: extract pixel-aspect-ratio information 2009-06-17 12:31:32 +02:00
Sebastian Dröge
fc9779bd86 matroskademux: Fix leaking of the Matroska TITLE element 2009-06-17 07:14:09 +02:00
Sebastian Dröge
55535c7fe6 effectv: Add basic documentation for the effectv elements 2009-06-16 20:38:42 +02:00
Sebastian Dröge
757f80b23e effectv: Define the fast PRNG function at a central place 2009-06-16 20:16:13 +02:00
Sebastian Dröge
414d0548e2 effectv: Move type definitions into separate headers
This is needed for the docs later.
2009-06-16 20:13:35 +02:00
Sebastian Dröge
44a822b29f effectv: Remove get_unit_size implementations
The default on from GstVideoFilter handles this already.
2009-06-16 19:41:02 +02:00
Wim Taymans
a0e3bc5997 udpsrc: add debug line for the socket 2009-06-16 15:06:50 +02:00
Ognyan Tonchev
375523be71 multiudpsink: add support for buffer lists
Add support for BufferList and add a unit test.

Fixes #585842
2009-06-16 15:04:15 +02:00
Wim Taymans
a2cef3e5c2 mp4gdepay: guess constantDuration better
Do a better job at guessing the constantDuration parameter when it is not
present in the caps.

Fixes #585205
2009-06-15 22:42:45 +02:00
Sebastian Dröge
cf58ba0fa5 warptv: Clean up warptv element and fix some minor bugs and leaks 2009-06-15 21:13:35 +02:00
Sebastian Dröge
f4fc26dc49 vertigotv: Clean up vertigotv element and fix some minor bugs and leaks 2009-06-15 21:13:35 +02:00
Sebastian Dröge
46ceaf957a dicetv: Use guint8 instead of char (which can be signed or unsigned) 2009-06-15 21:13:35 +02:00
Sebastian Dröge
aaf8c8769d shagadelictv: Use guint8/gint8 instead of char (which can be signed or unsigned) 2009-06-15 21:13:35 +02:00
Sebastian Dröge
1f65192185 shagadelictv: Clean up element and free all memory in finalize 2009-06-15 21:13:34 +02:00
Sebastian Dröge
f3dc59d010 revtv: Clean up revtv element 2009-06-15 21:13:34 +02:00
Sebastian Dröge
006caab053 quarktv: Simplify some code 2009-06-15 21:13:34 +02:00
Sebastian Dröge
438295acae quarktv: Use the input data if a NULL buffer is chosen instead of the value 0 2009-06-15 21:13:34 +02:00
Sebastian Dröge
426ff3bd93 quarktv: Fix setting the planes property of quarktv
Setting it to a value<16 would cause crashes before because
current_plane was set to the old number of planes-1. Also
fix calculations for non-2^n planes values.
2009-06-15 21:13:33 +02:00
Sebastian Dröge
c116c301af quarktv: Clean up the quarktv element 2009-06-15 21:13:33 +02:00
Sebastian Dröge
b3bb4fa7ef effectv: Make elements list constant 2009-06-15 21:13:33 +02:00
Sebastian Dröge
012f71afb5 edgetv: Clean up edgetv element and fix memory leak 2009-06-15 21:13:33 +02:00
Sebastian Dröge
0cbdfcbc22 dicetv: Clean up dicetv element and fix some smaller issues
This fixes a memory leak (the dice map) and a crash when
setting the square-bits property before caps are set.
2009-06-15 21:13:33 +02:00
Sebastian Dröge
d1ad31b9b3 agingtv: Actually use GstController for syncing the properties to timestamps 2009-06-15 21:13:32 +02:00
Sebastian Dröge
45ef8efc6b agingtv: Export some more agingtv properties via GObject properties 2009-06-15 21:13:32 +02:00
Sebastian Dröge
f8424d0166 agingtv: General cleanup and updating of copyright
Also make the scratch-lines property exported via a GObject
property and initialize/reset the internal state correctly.
2009-06-15 21:13:32 +02:00
Sebastian Dröge
865f11192c agingtv: Store and update state inside the instance struct
This makes the coloraging effect and pits effect visible.
2009-06-15 21:13:32 +02:00
Olivier Crête
4c782db81d rtpsirenpay: Remove deprecated symbol
Patch by: Luis Menina
2009-06-13 21:02:45 -04:00
Patrick Radizi
a95c049f76 rtspsrc: Add RTP blocksize functionality
Add property to make the client suggest a blocksize to the server.
Fixes #585549
2009-06-12 16:06:28 +02:00
Wim Taymans
6f3c1728f3 rtp: update README, fix some typos, mention gstrtpbin 2009-06-11 22:30:06 +02:00
Jan Schmidt
6d32642887 docs: Quieten a couple more docs warnings 2009-06-11 11:39:25 +01:00
Jan Schmidt
42c9f83b5f docs: Remove gtk-doc comment marker
These comment blocks aren't gtk-doc comments and cause annoying noise in
the docs build.
2009-06-11 11:27:26 +01:00
Sebastian Dröge
ceea8dc15f deinterlace: Implement upstream negotation 2009-06-11 10:06:04 +02:00
Sebastian Dröge
cd33adbd96 deinterlace: Improve debugging and clean up some code 2009-06-11 10:06:04 +02:00
Sebastian Dröge
101123a75d deinterlace: Clip buffers to the current segment if possible 2009-06-11 10:06:04 +02:00
Sebastian Dröge
4710a91cbf deinterlace: Clean up includes and clean up order of instance struct fields 2009-06-11 10:06:03 +02:00
Olivier Crête
645b9449a5 rtph263pay: Default to doing A, B and C modes, not only A 2009-06-10 16:09:56 -04:00
Sebastian Dröge
f297e37b5e deinterlace: Fix QoS calculations
The diff is a signed integer, not an unsigned one of course.

In modes other than GST_DEINTERLACE_ALL every frame has twice the
duration of the field duration.
2009-06-10 09:57:31 +02:00
Olivier Crête
dc61a71e5d rtpsirenpay: Put the bitrate in the RTP caps
The MS code seems to require the bitrate to interoperate and
draft-ietf-avt-rtp-g7221-00 also has it.
2009-06-09 14:13:31 -04:00
Sebastian Dröge
d0a5c0f3e3 deinterlace: Implement basic QoS
This change is based on Tim's QoS implementation
for jpegdec.
2009-06-09 19:56:53 +02:00
Sebastian Dröge
049722b680 deinterlace: Directly proxy events/queries to the peer pads
This removes some overhead introduced by the default handlers
that need to iterate over the other pads.
2009-06-09 19:56:53 +02:00
Edward Hervey
5e0a2296e0 avidemux: debug_memdump() unknown tags. Refactor junk parsing code.
This makes life slightly easier when debugging avi files.
2009-06-09 10:41:49 +02:00
Edward Hervey
6eec2a060d rtp: Don't forget to dist the headers for the CELT (de)payloaders. 2009-06-08 08:21:43 +02:00
Wim Taymans
9050fb26e9 Revert "Revert "qtdemux: fill timestamp table completely""
This reverts commit 9f022c8a85.

Sorry, I was thinking about the wrong module.
2009-06-07 20:54:06 +02:00
Wim Taymans
9f022c8a85 Revert "qtdemux: fill timestamp table completely"
This reverts commit 790b050fc5.
I forgot we were frozen.
2009-06-07 20:49:50 +02:00
Wim Taymans
790b050fc5 qtdemux: fill timestamp table completely
When there are less timestamps that there are samples, fill up the sample table
with the last know timestamp. This situation can happen when the last sample
does not decode and doesn't need a timestamp. We however calculate the total
track length using the last sample timestamp so we need to have something
sensible in there.

Fixes #585056
2009-06-07 20:46:45 +02:00
Wim Taymans
fb25aced92 wavparse: handle LIST INFO of 0 size
Handle LIST INFO chunks of 0 size instead of causing errors.

Fixes #584981
2009-06-07 13:37:04 +02:00
Wim Taymans
7a0c94121f Revert "wavparse: Remove dead assignments, move variable to where it's needed."
Reverts commit 44256a78f8 and use the result in
error reporting so that we can see what's going on.
2009-06-07 13:34:37 +02:00
Wim Taymans
ace2a84447 celtdepay: add CELT depayloader 2009-06-05 18:57:30 +02:00
Wim Taymans
ec554a7f02 rtpceltpay: add CELT RTP payloader 2009-06-05 18:57:30 +02:00
Sebastian Dröge
29111fa1f8 deinterlace: First try to handle DVD still frames correctly
This helps a bit with bug #582740 but still doesn't make it work.
2009-06-04 20:35:35 +02:00
Wim Taymans
b9ddf22340 rtspsrc: set the right state on rtpbin
We need to set the state of gstrtpbin to the same state as our source elements.
This fixes fallback to TCP again.
2009-06-04 15:19:05 +02:00
Wim Taymans
e7987ed06b vrawpay: trim output buffers
Remove the leftover unused bytes in the output buffer.
Fixes #584613
2009-06-02 19:33:28 +02:00
Wim Taymans
dabddb8126 vrawdepay: fix parsing of sampling field
commit a12d9a80f2 broke the parsing of the
sampling.
2009-06-02 19:33:28 +02:00