[gst-cvs] New commits on branch master
Tim Mueller
tpm at kemper.freedesktop.org
Tue Aug 11 01:33:39 PDT 2009
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=9590df3c2ec17c9dee0740aa605a591a10d8d94a
Author: Tim-Philipp Müller <tim.muller at collabora.co.uk>
Date: Tue Aug 11 03:08:01 2009 +0100
v4l2: fix make distcheck by disting some more headers
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=dbad1d424ab8511c01e31988de0b967c4ed5e233
Author: Tim-Philipp Müller <tim.muller at collabora.co.uk>
Date: Tue Aug 11 02:42:16 2009 +0100
docs: update
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=4701696a92664c1b88c7368441c73893e04698a8
Author: Tim-Philipp Müller <tim.muller at collabora.co.uk>
Date: Tue Aug 11 02:31:44 2009 +0100
Move rtpmanager from -bad to -good.
Hook up build infrastructure (autotools, docs, unit test).
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=92abe07e8011a9ac17d9fc9440fcf56106003dec
Author: ric <csxnju at sogou.com>
Date: Thu Aug 6 19:26:21 2009 +0200
rtpsource: avoid buffer leak on bad seqnum
Fixes #590797
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=9f68303a2e290c14c91deb5ec33abe885980010e
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Tue Jul 28 18:18:20 2009 +0200
rtpsource: allow for NULL caps on buffers
Add the NULL caps check where it matters and also cover another case of
potential NULL caps.
Fixes #590030
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e37844fdc7753fd41089f16c0b4cdc88f182bf68
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Tue Jul 28 11:59:56 2009 -0400
rtpsource: Incoming buffers do not always have caps
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=309113721796480c3cd9995d7d768cc6182c9f84
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon Jul 27 15:46:23 2009 +0200
rtpsession: avoid doing lip-sync in BYE
When we get a BYE packet, don't do lip-sync with the SR inside because some
senders have trouble constructing valid SR packets after BYE.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3747ede14a6909e11b19ae7363cd583f5ddb59ec
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon Jul 27 13:17:20 2009 +0200
rtpbin: don't do lip-sync after a BYE
After a BYE packet from a source, stop forwarding the SR packets for lip-sync
to rtpbin. Some senders don't update their SR packets correctly after sending a
BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
the current lip-sync instead.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d2ef095b804182f5b2f4a7a8a4d5a55088ac811d
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon Jul 27 12:43:02 2009 +0200
rtpbin: only reconsider once for BYE
When iterating the sources of a BYE packet, don't signal a reconsideration for
each of them but signal after we handled all sources.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e8c6bcdf8da3121b057121ba1a0904e588dc714e
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Tue Jul 21 15:33:41 2009 -0400
rtpsession: Free conflicting addresses on finalize
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=428368b44ae2ba9c488ddd8d87485b8a8ed1cdec
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Wed Jul 1 12:55:03 2009 +0200
rtpbin: use new method for netaddress to string
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=512ba93159d8c859ed515d0eb8c70a880d68a605
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon Jun 29 18:48:33 2009 +0200
rtpbin: do better cleanup of the src ghostpads
Connect to the pad-removed signal of the ptdemux elements so that we remove the
ghostpads for them. Fixes cleanup when going to NULL as well as when releasing
the sinkpads.
Fixes #561752
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d7a8663e055234ddad2c9cdc645279a733d1787e
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Thu May 28 19:08:40 2009 +0200
rtpsession: add a comment
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c53e595d23ae7ef8de82041f414c83f2108e7e4d
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon Jun 29 16:37:54 2009 +0200
rtpbin: add SDES property
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=9f330992f5548ea07821a02cb779a2da2c13c46b
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon Jun 29 16:21:05 2009 +0200
rtpbin: add SDES property that takes GstStructure
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d8496fb105d8733af2923dfc05f1d23466164043
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Tue Jun 2 17:46:08 2009 +0200
rtpbin: removed old gstrtpclient
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=779f67adc41f4ee99e9e4a2cd84cee2bef46e88a
Author: Branko Subasic <branko.subasic at axis.com>
Date: Fri Jun 19 19:09:19 2009 +0200
rtpbin: add support for buffer-list
Add support for sending buffer-lists.
Add unit test for testing that the buffer-list passed through rtpbin.
fixes #585839
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c5793a6a45cb05980a1a41301d51830bbf4ec4e0
Author: Tim-Philipp Müller <tim.muller at collabora.co.uk>
Date: Fri Jun 19 16:21:28 2009 +0100
Make build without warnings with debugging disabled
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=cf873498d2ec43c1205a2cadaf9011681942bbbb
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Thu May 28 17:37:44 2009 -0400
rtpbin: Transform the right session sdes message
Fixes #584165
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=dee142a945ac843b1ec602a3e2b8aa43315aa346
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Thu May 28 17:33:10 2009 -0400
Add ssrc to application/x-rtp-source-sdes structure
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=bf15048f4269176dd1cd5069c2d7dedf0a2a1007
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Wed May 27 11:03:14 2009 +0200
rtpsouce: the network address is in network order
Bring the network address in netowkr byte order to the host order.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=91eef691317bd6c44aefeaaa188e31ebdc297cec
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Tue May 26 15:40:52 2009 +0200
rtpsource: byteswap the port from GstNetAddress
Since the port in GstNetAddress is in network order we might need to byteswap it
before adding it to the source statistics.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=51251d0fa83dec0eb35d85ce059f1a2fa3dc3727
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon May 25 13:46:29 2009 +0200
rtpbin: remove ptdemux ghostpads
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=57653143c317627eef5c92ea2c2dd32f2799ea2c
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Mon May 25 13:33:20 2009 +0200
tests: add receive rtpbin unit test
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=7d9c2d20dfd02be8751bd4d711a9470d3ae19304
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 16:41:19 2009 +0200
rtpbin: add to new signal to remove SSRC pads
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=6c684e59c692bbe3e9ec02a66fcd03771de3fb41
Author: Ali Sabil <ali.sabil at gmail.com>
Date: Fri May 22 16:35:20 2009 +0200
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
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=48872d821544d3b22d59c337d1e44ac74f7a92b2
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 15:45:19 2009 +0200
rtpbin: use our ghostpads instead of its target
Since we keep a reference to our ghostpads, we can use them to track sessions.
This avoid us having to mess with the target of the ghostpad.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=02b34602c435758a2338207b0f8e24d3f0ccc574
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 15:37:29 2009 +0200
tests: more rtpbin checks
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=901b7f3b69c4d09789fd0a5ba2a62be1c0d5f0de
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 15:36:17 2009 +0200
rtpbin: don't warn when getting request pads twice
Allow getting the request pads multiple times, just return the previously
created pads.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=0ae6e3603b6b6774f0cab89f794ad0957210c852
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 13:47:30 2009 +0200
rtpsource: add RTP and RTCP source address
Add the RTP and RTCP sender addresses in the stats structure.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=62727e8fab630bf1abea36abb9f519c295758f17
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 13:45:15 2009 +0200
rtpsession: reuse source code for SDES
Reuse the RTPSource object property instead of duplicating code.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=232624c908b44ac33323844e4462dea98b83e0cb
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 13:44:17 2009 +0200
tests: add more rtpbin tests
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1710d8a3a51e377068d0e45d3061015bfb2d25b4
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 12:23:27 2009 +0200
tests: add rtpbin unit test
Add the beginnings of an rtpbin unit test
Add some more stuff to .gitignore
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1719af9113a93184111556f6beec72974071c897
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 12:20:13 2009 +0200
rtpbin: set target state on new elements
Set the state on newly added elements to the state of the parent.
Add some debug info and do some cleanups
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=9c92ee620931b6035158bbf9f12d4c62cbc0b41b
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 11:59:17 2009 +0200
rtpbin: unref requests pads after releasing
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a1c0bb24882327ee575e5de71bbab04c07f43bee
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Fri May 22 01:43:50 2009 +0200
rtpbin: Implement releasing the streams
See #561752
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e77542d3500f1eb94d647207eae107d33a9b832a
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Fri May 22 01:16:11 2009 +0200
rtpbin: Keep jb signals handler
Keep the signal handlers so they can be disconnected at release time
See #561752
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=59d0590cd764ccaf6d8898fae1745eac83a4cbd6
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri May 22 01:12:57 2009 +0200
rtpbin: use the right lock for the sessions
Use the right lock when iterating the sessions.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a9d6f3558c934f5f197c37146c1079062fcdf1fd
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Fri May 22 01:03:55 2009 +0200
rtpbin: Free session if request pads are released
Free the session when all the request pads are released.
Don't mess with the session list in free_session as it is called from a foreach
on that list.
Set the state of the upstream element to NULL first.
See #561752
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=46388b767fd7d8ef73da52999990ff1d722de28e
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Fri May 22 00:51:53 2009 +0200
rtpbin: Implement relasing of the rtp recv pad
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=35090984687ddbe0d2a6fb39838b895a7e3d5c11
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Fri May 22 00:44:51 2009 +0200
rtpbin: Implement releasing of rtp send pads
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2f6e9d7bf2008ce856b1b9ca0dd37dcbbced6064
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Fri May 22 00:34:36 2009 +0200
rtpbin: Implement release of the recv rtcp pad
See #561752
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=47d4bb90c1023b02f004182e27e6f4d96f7be520
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Fri May 22 00:16:19 2009 +0200
rtpbin: Implement releasing of rtcp src pad
See #561752
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=11607c4d638f7ae58ebdbbd532bad49d82a6e0f8
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Tue May 5 16:48:37 2009 +0200
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
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3482b47666d63eb85272314a95b645c68f4d7fbc
Author: Olivier Crete <olivier.crete at collabora.co.uk>
Date: Mon Apr 27 11:09:08 2009 +0200
rtpsouce: make WARNING into LOG
Since neither rtpmanager nor any of the payloaders properly implement
pad allocation, there is no way for the rtpmanager to inform downstream elements
of the new SSRC if there is an SSRC collision. So the warning is emitted all the
time and it is confusing.
Fixes #580144
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=63636b12907bd0c603cff9251bdba69ff1e0496a
Author: Olivier Crete <olivier.crete at collabora.co.uk>
Date: Mon Apr 27 11:06:01 2009 +0200
rtpsession: notify when SSRC changes
Emit a g_object_notify when the SSRc changes because of a collision.
Fixes #580144
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d45d18c735065ef13de3b7d55b1e4799f808e81d
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri Apr 17 16:16:29 2009 +0200
rtpsession: join the RTCP thread
Avoid a case where a joinable thread would be left unjoined, which leaked the
thread structure.
Fixes #577318.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=64046416ccbad20ee4a4627632e4d063dc3efcc8
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Wed Apr 15 18:14:48 2009 +0200
jitterbuffer: prevent overflow in EOS estimation
Use a guint64 instead of a guint to hold a 64bit value to prevent completely
bogues EOS estimation values due to overflows.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d6c623e90cb2fbbac4d87b039fd0d43fde1b16f2
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Wed Apr 15 17:44:17 2009 +0200
rtpbin: we should not provide a clock
There is no need to provide a clock.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=5ece6ae4e381367027d01a0ea04ca592cc87dc3d
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Wed Apr 15 17:28:56 2009 +0200
jitterbuffer: more estimated EOS fixes
Do more accurate EOS estimate and guard against backward timestamps.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=cbad89600c416f73ba0588e4e19919300541d18e
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Wed Apr 15 17:25:02 2009 +0200
jitterbuffer: release lock before pushing EOS
Make sure we release the jitterbuffer lock before we start pushing out data
because else we might deadlock.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=918c9448f2513543885d8e6effec5822b06c971e
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri Mar 27 17:44:57 2009 +0100
rtpbin: add on_npt_stop signal
Add the on_npt_stop signal to rtpbin and rtpjitterbuffer to notify the
application that the NPT stop position has been reached.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=55c3da71c1ffec1ea0cb7820ddc96ff9fda2b136
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri Mar 13 15:59:37 2009 +0100
rtpbin: don't return FALSE on seek events
Silently ignore the seek event instead of returning FALSE.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=109874ed50db0111b97ec11025199427f289d65a
Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date: Thu Feb 26 13:10:29 2009 +0100
gstrtpbin: Don't forward revc events to sender
Don't send events from the receiver to the sender side.
Fixes #572900.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=7ae3923ac669157dc50f5895519765a4d3a86d9e
Author: Stefan Kost <ensonic at users.sf.net>
Date: Wed Feb 25 11:45:05 2009 +0200
docs: various doc fixes
No short-desc as we have them in the element details.
Also keep things (Makefile.am and sections.txt) sorted.
Reword ambigous returns. No text after since please.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2c6ab34114fcc5af9602eb5b555a4e822e4e1994
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Fri Jan 23 12:13:00 2009 +0100
Send BYE packets immediatly for small sessions
When the number of participants is less than 50, the RFC allows for sending the
BYE packet immediatly instead of using the regular BYE timeout.
Fixes #567828.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=7f0b100db55ec912fb7d7c28530d402130055dbe
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Thu Jan 22 13:33:14 2009 +0100
Unlock the jitterbuffer before pushing out the packet-lost events.
Move some code before we do the unlock to make the jitterbuffer state
consistent while we are unlocked.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=dfdc9b66629f913571d181d4dc93ac35268e27b4
Author: Olivier Crete <tester at tester.ca>
Date: Fri Jan 2 17:40:06 2009 +0000
gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink):
* gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
When an SSRC is found on the caps of the sender RTP, use this as the
internal SSRC. Fixes #565910.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=0ad92e7da6fbfe24d3f6642ee9463cab029b4bb0
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Jan 2 16:50:53 2009 +0000
gst/rtpmanager/: Rename a method to better reflect what it really does.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_getcaps_send_rtp):
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_schedule_bye_locked), (rtp_session_schedule_bye):
* gst/rtpmanager/rtpsession.h:
Rename a method to better reflect what it really does.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=06d1532024cefe9791032104eb2362999ed2e47a
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 29 15:49:37 2008 +0000
gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp):
Use method to get the internal SSRC.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_set_property), (rtp_session_get_property):
Add property to congiure the internal SSRC of the session.
Fixes #565910.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1786eb1e253c1a21b413f13028745700efa9c1c6
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 29 15:21:58 2008 +0000
gst/rtpmanager/rtpsession.c: Only change the SSRC of the session and reset the internal source when the SSRC actually...
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
Only change the SSRC of the session and reset the internal source when
the SSRC actually changed. See #565910.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3fe87f7eab5982f2326e95d868d01fb1bc7a00f5
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 29 14:21:47 2008 +0000
gst/rtpmanager/rtpsource.*: When no payload was specified on the caps but there was a clock-rate, assume the clock-ra...
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate):
* gst/rtpmanager/rtpsource.h:
When no payload was specified on the caps but there was a clock-rate,
assume the clock-rate corresponds to the first payload type found in the
RTP packets. Fixes #565509.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2142edd3991bd9f41e4082588fa9a6f351e6b68a
Author: Arnout Vandecappelle <arnout at mind.be>
Date: Tue Dec 23 11:39:59 2008 +0000
gst/rtpmanager/rtpjitterbuffer.*: Keep track of the last outgoing timestamp and of the last sender-side time. Timest...
Original commit message from CVS:
Patch by: Arnout Vandecappelle <arnout at mind dot be>
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last outgoing timestamp and of the last sender-side
time. Timestamps can only go forward if they do at the sender
side, can only go back if they do at the sender side, and remain the
same if they remain the same at the sender side. Fixes #565319.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=5b6700a022aea3317a6792fa1ae400fa05eccece
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Nov 26 12:40:18 2008 +0000
gst/rtpmanager/rtpsession.c: Make obtain_source return an aditional ref so that we don't lose our ref to it when a se...
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (obtain_source),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye):
Make obtain_source return an aditional ref so that we don't lose our ref
to it when a session cleanup occurs when we are emiting a signal.
Emit the on_new_ssrc signal for the CSRC, not the SSRC.
Fixes #562319.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a80f7dc19aa5722ad84ad4a66ecf04abefb84bd2
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Nov 26 12:02:21 2008 +0000
gst/rtpmanager/gstrtpbin.c: Reset the sync parameters when clearing the payload type map too.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_reset_sync),
(gst_rtp_bin_clear_pt_map):
Reset the sync parameters when clearing the payload type map too.
Fixes #562312.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a2d7487ee192937b2dd7679d48c6130ac0e90447
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Nov 26 11:44:37 2008 +0000
gst/rtpmanager/gstrtpbin.*: Remove a lot of per stream state that is not needed and pass new info in the method call.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_client),
(gst_rtp_bin_reset_sync), (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream),
(gst_rtp_bin_class_init), (new_ssrc_pad_found):
* gst/rtpmanager/gstrtpbin.h:
Remove a lot of per stream state that is not needed and pass new info in
the method call.
Add signal to reset sync parameters.
Avoid parsing the caps to get a clock_base, we get this from the sync
signal now.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=b8408946b756091796797d38b6c0b8fd305e8249
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Nov 25 15:12:06 2008 +0000
gst/rtpmanager/gstrtpsession.c: Fix event leak.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtcp_src):
Fix event leak.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=ae346d9a6d8e5a9262fc8b7df7e9f9b9673ffb28
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Sat Nov 22 15:31:36 2008 +0000
gst/rtpmanager/rtpsession.c: Add property to configure the RTCP MTU.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_set_property),
(rtp_session_get_property):
Add property to configure the RTCP MTU.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=55bb4d5c955d37c229353896ac92a66d84093bd8
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Sat Nov 22 15:24:47 2008 +0000
gst/rtpmanager/rtpsession.c: Add G_PARAM_STATIC_STRINGS.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(copy_source), (rtp_session_create_sources),
(rtp_session_get_property):
Add G_PARAM_STATIC_STRINGS.
Add property to return a GValueArray of all known RTPSources in the
session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_create_sdes), (rtp_source_set_property),
(rtp_source_get_property):
Remove properties to set the various SDES items, an application is never
supposed to change the RTPSource data.
Change the SDES getter properties to one SDES property that returns all
SDES items in a GstStructure.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c84ffd84601fc7caef870d41cb37fc11177c4d5a
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Sat Nov 22 13:17:24 2008 +0000
gst/rtpmanager/gstrtpbin.c: Also unref the target pad for unknown pads.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
Also unref the target pad for unknown pads.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=75580396d95b2705c869337b4f13e737bdcc95d0
Author: Olivier Crete <tester at tester.ca>
Date: Fri Nov 21 16:17:22 2008 +0000
gst/rtpmanager/gstrtpbin.c: Release the right pads on rtpbin. Fixes #561752.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
Release the right pads on rtpbin. Fixes #561752.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2f5b130af3939199250c31fcb0f18c9034e2df6b
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Nov 20 18:41:34 2008 +0000
gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when processing RTP packets.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (get_current_times),
(rtcp_thread), (gst_rtp_session_chain_recv_rtp):
Pass the running time to the session when processing RTP packets.
Improve the time function to provide more info.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (update_arrival_stats),
(rtp_session_process_rtp), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (session_start_rtcp),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Mark the internal source with a flag.
Use running_time instead of the more useless timestamp.
Validate a source when a valid SDES has been received.
Pass the current system time when processing SR packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_create_stats),
(rtp_source_get_property), (rtp_source_send_rtp),
(rtp_source_process_rb), (rtp_source_get_new_rb),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add property to get source stats.
Mark params as STATIC_STRINGS.
Calculate the bitrate at the sender SSRC.
Avoid negative values in the round trip time calculations.
* gst/rtpmanager/rtpstats.h:
Update some docs and change some variable name to more closely reflect
what it contains.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e51423aab9bc756d7caa04975aa9f06cbb895643
Author: Sebastian Dröge <slomo at circular-chaos.org>
Date: Thu Nov 20 08:19:15 2008 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Initialize return value to fix compiler warning about uninitialized variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain_rtcp):
Initialize return value to fix compiler warning about uninitialized
variable.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d0ada6127e9f4e4d20c8989626455585bb21c726
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Nov 19 16:48:38 2008 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Mark signal arg as static scope.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init):
Mark signal arg as static scope.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=592c3f222f0947596b2a81bcbe1cba448925bae8
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Nov 19 09:06:29 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c3645239f58bb2e14110059e104338fc6d4fd44d
Author: Sebastian Dröge <slomo at circular-chaos.org>
Date: Mon Nov 17 19:47:32 2008 +0000
gst/rtpmanager/rtpsource.c: Fix GST_DEBUG call to only have as many arguments as required by the format string. Fixes...
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Fix GST_DEBUG call to only have as many arguments as required
by the format string. Fixes a compiler warning.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=5ab3e1059442f3bb77a26e982e834bd383faec47
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Nov 17 15:17:52 2008 +0000
gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ourselves but simply get the value from the ji...
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found):
Do not try to keep track of the clock-rate ourselves but simply get the
value from the jitterbuffer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add some debug info.
Pass the clock-rate to the jitterbuffer.
Also pass the clock-rate along with the rtp timestamp when getting the
sync parameters.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix some debug.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of clock-rate changes and return the clock-rate together with
the rtp timestamps used for sync.
Don't try to construct timestamps when we have no base_time.
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Request a new clock-rate when the payload type changes.
Reset the jitter calculation when the clock-rate changes.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1656fad93e59d9f6ac888539b3b8da3a2b318fc7
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Nov 13 15:48:54 2008 +0000
gst/rtpmanager/: Small cleanups and some more debug info.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain):
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew):
Small cleanups and some more debug info.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=6485d60a01c5468959c6a40a1a56df28cb19e121
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Nov 10 15:26:40 2008 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Also configure the next expected output seqnum when we get a seqnum-base on the ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
Also configure the next expected output seqnum when we get a seqnum-base
on the caps.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=b8352968094a7bf921a9c23fc837c79242a42da0
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Tue Nov 4 12:42:30 2008 +0000
Don't install static libs for plugins. Fixes #550851 for -bad.
Original commit message from CVS:
* ext/alsaspdif/Makefile.am:
* ext/amrwb/Makefile.am:
* ext/apexsink/Makefile.am:
* ext/arts/Makefile.am:
* ext/artsd/Makefile.am:
* ext/audiofile/Makefile.am:
* ext/audioresample/Makefile.am:
* ext/bz2/Makefile.am:
* ext/cdaudio/Makefile.am:
* ext/celt/Makefile.am:
* ext/dc1394/Makefile.am:
* ext/dirac/Makefile.am:
* ext/directfb/Makefile.am:
* ext/divx/Makefile.am:
* ext/dts/Makefile.am:
* ext/faac/Makefile.am:
* ext/faad/Makefile.am:
* ext/gsm/Makefile.am:
* ext/hermes/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/jack/Makefile.am:
* ext/jp2k/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/lcs/Makefile.am:
* ext/libfame/Makefile.am:
* ext/libmms/Makefile.am:
* ext/metadata/Makefile.am:
* ext/mpeg2enc/Makefile.am:
* ext/mplex/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/mythtv/Makefile.am:
* ext/nas/Makefile.am:
* ext/neon/Makefile.am:
* ext/ofa/Makefile.am:
* ext/polyp/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/sdl/Makefile.am:
* ext/shout/Makefile.am:
* ext/snapshot/Makefile.am:
* ext/sndfile/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/spc/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/tarkin/Makefile.am:
* ext/theora/Makefile.am:
* ext/timidity/Makefile.am:
* ext/twolame/Makefile.am:
* ext/x264/Makefile.am:
* ext/xine/Makefile.am:
* ext/xvid/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/dshow/Makefile.am:
* gst/aiffparse/Makefile.am:
* gst/app/Makefile.am:
* gst/audiobuffer/Makefile.am:
* gst/bayer/Makefile.am:
* gst/cdxaparse/Makefile.am:
* gst/chart/Makefile.am:
* gst/colorspace/Makefile.am:
* gst/dccp/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/deinterlace2/Makefile.am:
* gst/dvdspu/Makefile.am:
* gst/festival/Makefile.am:
* gst/filter/Makefile.am:
* gst/flacparse/Makefile.am:
* gst/flv/Makefile.am:
* gst/games/Makefile.am:
* gst/h264parse/Makefile.am:
* gst/librfb/Makefile.am:
* gst/mixmatrix/Makefile.am:
* gst/modplug/Makefile.am:
* gst/mpeg1sys/Makefile.am:
* gst/mpeg4videoparse/Makefile.am:
* gst/mpegdemux/Makefile.am:
* gst/mpegtsmux/Makefile.am:
* gst/mpegvideoparse/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/nuvdemux/Makefile.am:
* gst/overlay/Makefile.am:
* gst/passthrough/Makefile.am:
* gst/pcapparse/Makefile.am:
* gst/playondemand/Makefile.am:
* gst/rawparse/Makefile.am:
* gst/real/Makefile.am:
* gst/rtjpeg/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/scaletempo/Makefile.am:
* gst/sdp/Makefile.am:
* gst/selector/Makefile.am:
* gst/smooth/Makefile.am:
* gst/smoothwave/Makefile.am:
* gst/speed/Makefile.am:
* gst/speexresample/Makefile.am:
* gst/stereo/Makefile.am:
* gst/subenc/Makefile.am:
* gst/tta/Makefile.am:
* gst/vbidec/Makefile.am:
* gst/videodrop/Makefile.am:
* gst/videosignal/Makefile.am:
* gst/virtualdub/Makefile.am:
* gst/vmnc/Makefile.am:
* gst/y4m/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/cdrom/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
* sys/dvb/Makefile.am:
* sys/dxr3/Makefile.am:
* sys/fbdev/Makefile.am:
* sys/oss4/Makefile.am:
* sys/qcam/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/vcd/Makefile.am:
* sys/wininet/Makefile.am:
* win32/common/config.h:
Don't install static libs for plugins. Fixes #550851 for -bad.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=eaa23fd49a45aa7b124db513ef301254fca7e405
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Oct 16 13:05:37 2008 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Fix problem with using the output seqnum counter to check for input seqnum disco...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix problem with using the output seqnum counter to check for input
seqnum discontinuities.
Improve gap detection and recovery, reset and flush the jitterbuffer on
seqnum restart. Fixes #556520.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert):
Fix wrong G_LIKELY.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3563bbaabd5ef91346f574156cf3e892e46ecb8e
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Oct 16 09:51:28 2008 +0000
gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, make LATENCY event return
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src):
Install event handler on the rtcp_src pad, make LATENCY event return
TRUE.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3bebd53b6fcdf53d6d5fc5d2e7281c7e11fe8300
Author: HÃ¥vard Graff <havard.graff at tandberg.com>
Date: Tue Oct 7 18:54:41 2008 +0000
gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.
Original commit message from CVS:
Patch by: HÃ¥vard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin-marshal.list:
Add marshaller for new action signal.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add action signal to retrieve the internal RTPSession object.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_get_property), (gst_rtp_session_release_pad):
Add property to access the internal RTPSession object.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(check_collision):
* gst/rtpmanager/rtpsession.h:
Add action signal to retrieve an RTPSource object by SSRC.
See #555396.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=bd8f4b6c58df1bdb9d412ca11b164196d731fb15
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Oct 7 11:33:10 2008 +0000
gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_pad),
(free_session), (gst_rtp_bin_dispose), (remove_recv_rtp),
(remove_recv_rtcp), (remove_send_rtp), (remove_rtcp),
(gst_rtp_bin_release_pad):
Release pads of the session manager.
Start implementing releasing pads of gstrtpbin.
* gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink),
(remove_recv_rtcp_sink), (remove_send_rtp_sink),
(remove_send_rtcp_src), (gst_rtp_session_release_pad):
Implement releasing pads in gstrtpsession.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=4553863755c5657f0042697328b1f29268fdb5bd
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Oct 7 10:02:20 2008 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Only update the seqnum-base when it was not already configured for the streams.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps):
Only update the seqnum-base when it was not already configured for the
streams.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=55b7860cc4dbdf591dbe2543f3a5be946b027a97
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Sep 30 15:08:52 2008 +0000
gst/rtpmanager/rtpsession.c: Ref the rtpsource object before we release the session lock when we emit the signals.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes),
(on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout):
Ref the rtpsource object before we release the session lock when we emit
the signals.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c2c69bfb865d2cf0da10204514edf34d7d935025
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Sep 23 18:13:31 2008 +0000
gst/rtpmanager/: Fix some docs.
Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpsession.c: (on_sender_timeout),
(session_cleanup):
* gst/rtpmanager/rtpsource.c:
Fix some docs.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a2b86bbce59fab0cdee49744bbc1a3594a037c21
Author: Jan Schmidt <thaytan at mad.scientist.com>
Date: Wed Sep 17 13:59:21 2008 +0000
Fix compiler warnings on OS/X
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_process_cb):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Fix compiler warnings on OS/X
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=5e98fa572fa7c6e9a1dfbc51c53d73c4241aa2ee
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Sat Sep 13 01:37:50 2008 +0000
gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A...
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
Do not try to adjust the offset of streams for which we have not yet
seen an SR packet. Avoids large ts-offsets in some cases.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=85e26f65468b6407ef753220c70695ef87700045
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Sep 5 13:52:34 2008 +0000
gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=5c89bb2ab3a15eafcb59581cfc2bf5e477cafc73
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Aug 28 15:21:45 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=62ecaee7488a18fb71ef012b0161a227576d1e87
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Aug 13 14:31:02 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=cc74738d8331d9cf8ea58754dec3d87f0692e938
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Mon Aug 11 07:20:15 2008 +0000
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?
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d392defbd3c060b7a32cd9720635e919586fefce
Author: Olivier Crete <tester at tester.ca>
Date: Tue Aug 5 09:42:53 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1bef5a8ab8f65f05fcb20d77d8d1192d1e27e85f
Author: HÃ¥vard Graff <havard.graff at tandberg.com>
Date: Tue Aug 5 09:00:50 2008 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum.
Original commit message from CVS:
Patch by: HÃ¥vard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Fix debug by logging the right seqnum.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2707a84d7816dd5e180980586e7315a295ab074d
Author: Olivier Crete <tester at tester.ca>
Date: Tue Aug 5 08:58:27 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=fd44690d4fc9bf4eaeb261503fcd7fe0dc9be59f
Author: Peter Kjellerstedt <pkj at axis.com>
Date: Thu Jul 3 14:44:51 2008 +0000
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).
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e2f49d9ccf103fdd25212ce3c261344dc47202b7
Author: Peter Kjellerstedt <pkj at axis.com>
Date: Thu Jul 3 14:31:10 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=ca15984e14415a6be286c4744020aa4f9b258979
Author: Peter Kjellerstedt <pkj at axis.com>
Date: Thu Jul 3 13:47:19 2008 +0000
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().
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a71ffc55d8477f7bcf52f80fbffaf0991dafc263
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Mon Jun 16 07:30:34 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=138c2b7cf9b713b5d6e56ac7934e228703bfc9f5
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Mon Jun 16 07:03:58 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2d1ccbf52e310773437f5daa23c29c4ee0488334
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Thu Jun 12 14:49:18 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=8dc879f15e24e4dc1145547e70e5b0d004f81437
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Jun 6 13:01:05 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=fda8195d763bddf36b6e4255a921bf8a09421f2a
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue May 27 16:48:10 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=bd1e0ebfc03c9c74492fc73533fe783b4795dd01
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon May 26 10:09:29 2008 +0000
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..
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=b889dfad303ccba1bcedbd67cd85a127344ee20e
Author: HÃ¥vard Graff <havard.graff at tandberg.com>
Date: Mon May 26 10:00:24 2008 +0000
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: HÃ¥vard 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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=6716231857c4f6e0c81bfc7fd94409aac110950f
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon May 26 09:57:40 2008 +0000
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().
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=4e5347c8feb5583ed3d806a7d51a20ab2b1da05a
Author: Jan Schmidt <thaytan at mad.scientist.com>
Date: Fri May 16 19:56:30 2008 +0000
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
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2506d13ecc0799000634e7190a4ff526b7ab5980
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed May 14 21:02:19 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=cd00eb71b42873ece10c2af6f7021c8b7fa229bb
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue May 13 09:06:51 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=71c25106654b61122e9a3781fb7da9131a965567
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon May 12 18:43:41 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=fd8061784a013efc876cf0dcd45998fcc11582e3
Author: Peter Kjellerstedt <pkj at axis.com>
Date: Mon May 12 14:28:09 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=95ab28208341f7e1f21f4701a6e9c98df9c7479e
Author: Jan Schmidt <thaytan at mad.scientist.com>
Date: Mon May 12 14:12:08 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=b1ef03968a6d488c9ec88b70dad8fcf3301f958c
Author: Peter Kjellerstedt <pkj at axis.com>
Date: Fri May 9 07:41:58 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=bddddbd409f33dfeb94bfcf14e3a01e19a6fd433
Author: Olivier Crete <tester at tester.ca>
Date: Thu May 8 09:43:33 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c466ae6bdc7ccb4c93f07f1952d4d1221fa6bf59
Author: Sjoerd Simons <sjoerd at luon.net>
Date: Thu May 8 06:23:39 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=d6c880973993ab98611186f309e7e957062f8956
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 25 11:32:09 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=250c38a5ce174198ce6007d65f2ef8c1a0d020c4
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 25 11:22:13 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e2ab966d145747d6ec97a5ea3558250e8826a2fd
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 25 09:35:43 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a05b42ef04802d78164bd345040052a900d690a4
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 25 08:21:06 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e779adca6947da83100a9af502c36d021882b117
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 25 08:15:58 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3c5cf0cd38ac558736197d73bd2a9c0f04f7958c
Author: Olivier Crete <tester at tester.ca>
Date: Mon Apr 21 08:26:37 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=4cc70a0c227836522cf018e13b9dfb537331baa2
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Apr 17 07:31:44 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=959c341cbda42c05aaa8b3bb4b1712c99205a7fc
Author: Peter Kjellerstedt <pkj at axis.com>
Date: Wed Apr 9 22:27:50 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3f5884708033ba3579e24b22ce3bb44657eec66f
Author: Olivier Crete <tester at tester.ca>
Date: Tue Mar 11 12:40:58 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=6ba2fcd4ffc3d84755ef953eaa1494fff76fcc67
Author: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg.com>
Date: Tue Mar 11 11:36:03 2008 +0000
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 at 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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=52cdd3c59a5cd4d9b48c379a5e7227407e5c97cb
Author: Stefan Kost <ensonic at users.sf.net>
Date: Tue Mar 11 10:43:32 2008 +0000
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 at users.sf.net>
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Don't try to reset the clock skew when we have no timestamps.
Fixes #519005.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=db8bdc8b927ef5156bbfc84c9c084abc8b667822
Author: Olivier Crete <tester at tester.ca>
Date: Wed Feb 20 09:33:25 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a301c9a22b5c442ec072fa9c2fc298f59a9ab21b
Author: Olivier Crete <tester at tester.ca>
Date: Thu Feb 14 16:25:51 2008 +0000
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 at tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate):
Ignore streams that did not receive an SR packet when doing
synchronisation. Fixes #516160.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=b63862605311a9502816255297fc8254c6478029
Author: Thijs Vermeir <thijsvermeir at gmail.com>
Date: Tue Jan 29 18:57:27 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=7b2446b67665a688029bd8776d950b9836d5664f
Author: Olivier Crete <tester at tester.ca>
Date: Fri Jan 25 16:58:00 2008 +0000
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 at 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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=41ada27f2e279e88d3d3e70f02d34bf010a5d0e2
Author: Olivier Crete <tester at tester.ca>
Date: Fri Jan 25 16:00:52 2008 +0000
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 at 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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=eb0993af12fdbaa5182fdc13cc7d20cf34af2c22
Author: Olivier Crete <tester at tester.ca>
Date: Fri Jan 25 15:49:55 2008 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.
Original commit message from CVS:
Patch by: Olivier Crete <tester at 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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=0369f87020824397c7b176a139a63cc708f91cf5
Author: Olivier Crete <tester at tester.ca>
Date: Fri Jan 25 01:44:27 2008 +0000
gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. Fixes #511920
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester.ca>
* gst/rtpmanager/rtpsource.c: Fix unref of buffer using the
wrong function. Fixes #511920
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=6e6c59a198d7c04c54cbf990b7f1491042c05c7f
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Jan 11 17:02:30 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=03d9faf5fa8e27f433d5c247801ec7257a7428f7
Author: Youness Alaoui <youness.alaoui at collabora.co.uk>
Date: Fri Jan 11 16:45:57 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c6d892420a848ff3d0fcfd73ff4f176a966c77a4
Author: Thijs Vermeir <thijsvermeir at gmail.com>
Date: Thu Jan 10 20:57:17 2008 +0000
gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Fix documentation for latest patch
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a4db9d0943bba023d39d7817f20146a9b4792b63
Author: Thijs Vermeir <thijsvermeir at gmail.com>
Date: Thu Jan 10 14:34:30 2008 +0000
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)
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c7818b0c0fa84247eda94b777d634aeff7ef824b
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Jan 9 14:39:44 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c5e9700edaa15145b2c31cef77ec870c1d96789f
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Jan 4 18:47:57 2008 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=cba910a4300774d621dc341db81a24f744795ba8
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 31 13:12:06 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a965ebff09d986440c340b3d609e45579fe74903
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Dec 12 16:59:03 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=df55cf2f087521bb55426669025ab45682aa9972
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Dec 12 12:11:53 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=771ed2339d87d2daf79abbafd54b705d129dd159
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 10 18:36:04 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=49e501a64731c016f3f1f49136981850f6ec87a8
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 10 15:34:19 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=95d1f62397fab2ef07c4f55af04c9cc10e9533b0
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 10 14:03:32 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1971ae0d82153aa2d1cc65c68f1fcd4538fcaf70
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Dec 10 11:08:11 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1a8f489093c80935c4d9fc253ae8c50079d04380
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Nov 22 09:08:27 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c5fdb6bff39b087ae17ed20cec9fa87815eaf465
Author: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg.com>
Date: Fri Nov 2 21:45:38 2007 +0000
gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg.com>
* gst/rtpmanager/gstrtpsession.c:
Fix bad function signatures (#492798).
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=8da59edc68158a45555c141b54dc4ac3a434b770
Author: Laurent Glayal <spglegle at yahoo.fr>
Date: Tue Oct 9 10:01:39 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c924d4a466de1178100f16ea2c3322316c2d7016
Author: Jan Schmidt <thaytan at mad.scientist.com>
Date: Mon Oct 8 17:46:45 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=4556ccb666b0179a5adb3eddcaf28278f565250f
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Oct 8 10:39:35 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=76a89b5e500952afb6d8d0c544848c7c99bb0427
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Oct 5 17:26:14 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=387f41e157e0aac82d798152765e1c952c3a7c47
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Oct 5 12:07:37 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=b09507ab0c1aff1b169709b3da4b42107532335f
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Oct 2 10:27:45 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=9c867a2160632d541397646081cfdf9be3818499
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Sep 28 14:51:58 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2b1f49a26e1879b41329e9198f65826fdce960a4
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Sep 28 11:17:35 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=fa00695a390e507a049bca04446ed65c69c6641b
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Sep 26 20:08:28 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=949f1685ce1c3ddf04873825f1bdc1c0bb7f284e
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Sep 20 14:34:57 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=56d583228740f37745db800a010b3f7b4ce6f164
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Sep 17 02:01:41 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=b2aa36cb0d809341ab81e8a146be3a8bb06f2c1b
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Sun Sep 16 19:40:31 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=0441ef80b0fbe22b0a6270958e4ad2058dc0d423
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Sat Sep 15 18:48:03 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a93348cc6da0eff1a30fd08b2ec17962079a38a6
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Sep 12 21:23:47 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=919deb449063c2b26485b7c07b829e081b7c7fae
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Sep 12 18:04:32 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=aa8985d1e40d4d847788ff45b76fedba0e9cd6f5
Author: Tim-Philipp Müller <tim at centricular.net>
Date: Tue Sep 4 15:23:34 2007 +0000
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).
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e7b6212c51e5185023dde6170c3de0b975c134d7
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Sep 3 21:19:34 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=f4e6f223159f4e2041fcde2a72014e772ff148a4
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Aug 31 15:26:14 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=c576bcec15da40d57c674dce57675ed5e3a93e5b
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Aug 29 16:56:27 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=325dac0fc21bf3e014f1a0bbe6de5dc711e21e30
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Aug 29 01:22:43 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=eb86865a62b01688fcfdbcb2c0d374e36d37ec3f
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Aug 28 20:30:16 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=6835b966ec20faaea3a81d6486f8378c884fa5c4
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Aug 27 21:17:21 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=10d6ba4d6163a6e8e10cc302a5d114098ab538cf
Author: Tim-Philipp Müller <tim at centricular.net>
Date: Thu Aug 23 21:39:58 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=f13ad91c77218f69ac5256632607ab6d3f087874
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Aug 21 17:18:29 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=ce70e0f43e12a4efbdf7e92ce223005ae41386c4
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Aug 21 16:04:47 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=076da98efb18fd7df3020baed7ead7f22ef0f219
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Aug 16 11:40:16 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=f24c54f4b55f67c6c5e4488ed18a809d01390b7e
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Mon Aug 13 06:16:40 2007 +0000
gst/rtpmanager/rtpjitterbuffer.c: Include stdlib.
Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c:
Include stdlib.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=cdd82f2a95855f0a3cc0b7be2f1b69d99eaba729
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Aug 10 17:16:53 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=366a7565525789adedcef38236e7ee81b590c981
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Wed Jul 18 07:35:32 2007 +0000
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).
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=31571c8cb2c943ca135b7257df80b04ef5c1edee
Author: Jens Granseuer <jensgr at gmx.net>
Date: Fri Jun 22 20:23:18 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=0c4fe985b638ee046102e328b66b087d64f68679
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon May 28 16:37:47 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2a8cfc6410cf29f58287d4ad985e4381e9ff6c61
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed May 23 13:08:52 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=3bc059707de50fc6f5e9d100cb5f4094d6ca30b7
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue May 15 13:29:53 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=720dfeb3a57ba6813f159e623aafa5b3b1fb0f43
Author: David Schleef <ds at schleef.org>
Date: Tue May 15 03:45:45 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=62d401eb93ab109c0fed8e0133012bd5cc0ae5de
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon May 14 15:28:36 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=15b54ec7e2160b83fd79f6efc70034427ab43a08
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Thu May 10 14:02:07 2007 +0000
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 at gmail.com>).
* gst/switch/Makefile.am:
Add require libraries(spotted by Ali Sabil <ali.sabil at gmail.com>).
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=091c2cfbc00f9c6b0085eac3266a31da39e0e20c
Author: Stefan Kost <ensonic at users.sourceforge.net>
Date: Thu May 10 12:38:49 2007 +0000
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 at gmail.com>)
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=88f24417222ba7d627587cca545dae9079631c27
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed May 9 11:24:22 2007 +0000
gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Pass queries upstream.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a241c62ecb8beebb20621596a40d2b1a7fdc44d9
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri May 4 12:32:27 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=600afaaff9e489c9e3ea3d5546fd58568564be40
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Mon Apr 30 13:41:30 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=e6537bcd7c8b00d8fc1236c4fd0bd0a22cb5537c
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Sun Apr 29 14:46:27 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a7b80281d1af025bde9fba40c5eee798a00f2f21
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 27 15:09:12 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=43f0b878c9acd2314bf9a9f260e15ef77169e778
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Apr 25 16:38:03 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=333764307d21eb430790c848a40f30521c34ffab
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Apr 25 15:48:46 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=ae536e0c8920ca75190d81ec45894f12faceb3c2
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Apr 25 13:19:36 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=23883be047f94d8bde7094b6210277e92e9df11c
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Apr 25 08:30:48 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=677b361dc33b4a08fa18df88a64f4d49a511647d
Author: Tim-Philipp Müller <tim at centricular.net>
Date: Sat Apr 21 19:21:49 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=54b3dec1f53c823bd947685bf89c4d0e041c2e2a
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Apr 18 18:58:53 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=490113d40db4fc3c291501941a06b3846ace1bb2
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 13 09:20:55 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=8bbea77a41c38fe63172fd511cdd8632d577fd72
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Apr 12 08:18:32 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=03bf43d50edb50f8114419cd62278afd7f67922b
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Apr 11 13:49:54 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=8c67b5d7dd1457aa0bafc89fbd022a0a6d79088c
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Apr 10 09:14:07 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a6aa41dc2118c811a5e84246722e8c6e99114c32
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 6 12:28:29 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1b0ae2608f93a9016abd77bdc79db93f0ad58ad8
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Fri Apr 6 12:07:30 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=63dbc757340c20c5fb20e622cd210e7957cfadb4
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Apr 5 16:10:24 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=9bfc641f0d4bb08feba724a860506e775937ead1
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Thu Apr 5 13:54:23 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a9d14ed31036aacb46cf97b3ced0bf9cd8cb434b
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Wed Apr 4 10:23:15 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=5351f0cb519f65352d8d79540ee16da3871705b0
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Apr 3 11:35:39 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=fdae491de7e2c014e62560c03a4f5ca40ded9368
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Apr 3 09:51:13 2007 +0000
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.
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=f0d1ab1c1f680f18a2ef9ec09681ad77829e45b1
Author: Wim Taymans <wim.taymans at gmail.com>
Date: Tue Apr 3 09:13:17 2007 +0000
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.
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