Commit graph

55 commits

Author SHA1 Message Date
Olivier Crête
b3f5cdd1f9 rtpssrcdemux: Forward sticky events to new pads 2012-01-27 19:05:24 +01:00
Olivier Crête
76c93af537 rtpssrcdemux: Add ssrc to forwarded CAPS events
Also iterate the list of GstRtpSsrcDemuxPad safely
2012-01-27 19:05:23 +01:00
Olivier Crête
3285c45dbc rtpssrccdemux: Factor out getting dpad by pad 2012-01-27 19:05:23 +01:00
Wim Taymans
1584806634 port to new gthread API 2012-01-19 11:33:53 +01:00
Sebastian Dröge
93e3ed5a86 Merge branch 'master' into 0.11
Conflicts:
	ext/cairo/gsttextoverlay.c
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstaacparse.c
	gst/avi/gstavimux.c
	gst/flv/gstflvmux.c
	gst/interleave/interleave.c
	gst/isomp4/gstqtmux.c
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-mux.h
	gst/matroska/matroska-read-common.c
	gst/multifile/gstmultifilesink.c
	gst/multipart/multipartmux.c
	gst/shapewipe/gstshapewipe.c
	gst/smpte/gstsmpte.c
	gst/udp/gstmultiudpsink.c
	gst/videobox/gstvideobox.c
	gst/videocrop/gstaspectratiocrop.c
	gst/videomixer/videomixer.c
	gst/videomixer/videomixer2.c
	gst/wavparse/gstwavparse.c
	po/ja.po
	po/lv.po
	po/sr.po
	tests/check/Makefile.am
	tests/check/elements/qtmux.c
	tests/check/elements/rgvolume.c
2012-01-10 14:32:32 +01:00
Tim-Philipp Müller
b8b8454bcb Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
2011-12-12 09:46:27 +00:00
Wim Taymans
9e27b122d9 ssrcdemux: fix iterator and caps 2011-12-10 11:13:38 +01:00
Wim Taymans
439e2f1cfd rtp: fix marshallers
Remove custom marshallers for minobject.
Init RTCP buffer correctly.
Handle results from setcaps
Remove asserts.
2011-12-09 10:51:14 +01:00
Edward Hervey
86a57e3546 rtpmanager: Initialize GstRTPBuffer before usage 2011-12-05 18:40:12 +01:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Wim Taymans
7cc4b72550 add parent to internal links 2011-11-16 17:54:49 +01:00
Wim Taymans
6190312214 add parent to query function 2011-11-16 17:27:13 +01:00
Wim Taymans
a19a4a69ae more template fixes 2011-11-04 13:12:37 +01:00
Edward Hervey
d4a2a46606 rtpssrcdemux: Fix wrong usage of gst_iterator_filter
It takes a GValue* as the user_data.

And don't forget to unref the demuxer before returning.
2011-10-13 09:34:04 +02:00
Wim Taymans
4121021bb2 Merge branch 'master' into 0.11
Conflicts:
	ext/pulse/pulsesink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtp/gstrtph264pay.c
	gst/rtpmanager/gstrtpssrcdemux.c
2011-08-03 18:25:30 +02:00
Mark Nauwelaerts
3a98f6f0fd rtpssrcdemux: keep a ref on the src pad while using it
Prevent a possible race if clear_ssrc() is called between getting the pad and
doing the push.

Based on patch by <olivier.crete@collabora.com>

https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:51:01 +02:00
Olivier Crête
c7b9b98648 rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit
We need to keep the lock held because we don't want a push before the "new-ssrc-pad"
handler has completed. But we may want to push an event from inside that handler, hence
the recursive mutex.

https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:50:59 +02:00
Olivier Crête
e26b5391c2 rtpssrcdemux: Use PADs lock
https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:50:57 +02:00
Mark Nauwelaerts
ef02634dc6 rtpmanager: port to 0.11
* use G_DEFINE_TYPE
* adjust to new GstBuffer and corresponding rtp and rtcp buffer interfaces
* misc caps and segment handling changes

FIXME: also relies on being able to pass caps along with a buffer,
which has no evident equivalent yet, so that either needs one,
or still needs quite some code path modification to drag along caps.
2011-07-06 10:16:12 +02:00
Sebastian Dröge
b694bfeca3 ssrcdemux: Fix uninitialized variable compiler warning for (pre-) releases too 2011-05-17 10:47:32 +02:00
Sebastian Dröge
0f05d3e5a5 rtpssrcdemux: Fix uninitialized variable compiler warning 2011-05-17 09:24:08 +02:00
Olivier Crête
b6bfc512e8 ssrcdemux: Implement iterate internal links for sink pads
https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:29 +02:00
Olivier Crête
23b6c8febc rtpssrcdemux: iterate pad function is only valid for src pads
The iterate function is only used for src pads, so mark it as such and remove
dead code.

https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:25 +02:00
Olivier Crête
1bf94a92b0 rtpssrcdemux: Release lock before emitting signal
If the lock is not released before emitting a signal, it may cause a deadlock
if any other function in the element is called.

Also removed an unused timestamp parameter

https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:20 +02:00
Pascal Buhler
0d2d52856f rtpssrcdemux: Unknown SSRC is not fatal
https://bugzilla.gnome.org/show_bug.cgi?id=646966
2011-04-11 17:37:58 -04:00
Sebastian Dröge
31af4fe33e rtp: Unref events if the parent element disappeared 2011-04-08 15:20:51 +02:00
Ole André Vadla Ravnås
046f170d6a rtpmanager: fix pad callbacks so they handle when parent goes away
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.

This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
2011-04-08 15:16:56 +02:00
Olivier Crete
c7b1ce7310 rtpssrcdemux: Tag upstream custom events with SSRC 2011-02-01 16:49:10 +01:00
Mark Nauwelaerts
46c91476eb rtpssrcdemux: do not hold custom PAD_LOCK when pushing downstream 2010-12-03 15:50:21 +01:00
Sebastian Dröge
f16ed4a91c gst: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs 2010-06-06 17:52:40 +02:00
Benjamin Otte
cccfeaa59c gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:32:00 +01:00
Wim Taymans
c611bbaa8e ssrcdemux: fix comparison
A NULL means no pad was found.
2009-12-21 15:07:34 +01:00
Sebastian Dröge
000a483d31 rtp: Use new gst_iterator_new_single() for the internal linked pads iteration 2009-08-31 08:09:09 +02:00
Sebastian Dröge
97cb7bdb6c rtpssrcdemux: Use iterate internal links instead of deprecated get internal links 2009-08-31 08:09:08 +02:00
Ali Sabil
6c684e59c6 ssrcdemux: emit signal when pads are removed
Add action signal to clear an SSRC in the ssrc demuxer.
Add signal to notify of removed ssrc.

See #554839
2009-08-11 02:30:44 +01:00
Wim Taymans
11607c4d63 rtpssrcdemux: drop unexpected RTCP packets
We usually only get SR packets in our chain function but if an invalid packet
contains the SR packet after the RR packet, we must not fail but simply ignore
the malformed packet.

Fixes #581375
2009-08-11 02:30:41 +01:00
Wim Taymans
592c3f222f gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to get lip-sync notifications.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream), (free_stream),
(new_ssrc_pad_found):
Remove internal sync pad, use signals instead to get lip-sync
notifications.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
(remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
(gst_rtp_jitter_buffer_release_pad),
(gst_rtp_jitter_buffer_sink_rtcp_event),
(gst_rtp_jitter_buffer_chain_rtcp),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Make it possible to send SR packets to the jitterbuffer.
Check if the SR timestamps are valid by comparing them to the RTP
timestamps.
Signal the SR packet and the timing information to listeners.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
Remove some unused code.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last seen RTP timestamp so that we can filter out
invalid SR packets.
2009-08-11 02:30:38 +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
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
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
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
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
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
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
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
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