+=== release 0.10.16 ===
+
+2009-08-29 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ releasing 0.10.16, "Secret Handshakes"
+
+2009-08-26 00:58:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ 0.10.15.5 pre-release
+
+2009-08-25 16:53:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: don't use relative seeks
+ Don't use relative seeks, it's too hard to track where we are after a flush
+ etc.
+ fixes #593015
+
+2009-08-24 17:50:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ 0.10.15.4 pre-release
+
+2009-08-24 16:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: don't discard the result of _set_caps()
+ Use the result of gst_pad_set_caps() instead of assuming success.
+ See #590678
+
+2009-08-21 11:44:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_fourcc.h:
+ qtdemux: add support for agsm
+ Fixes #592530
+
+2009-08-18 17:16:11 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix qt style string tag extraction
+ QT style tags are tested on starting with (C) symbol using >>,
+ and (unsigned) int (may) have different >> behaviour.
+ Fixes #592232.
+
+2009-08-17 15:48:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/jpeg/smokecodec.c:
+ smokeenc: don't crash when compiled against libjpeg7
+ Set parameters so that we don't crash with libjpeg7. Based on
+ Stefan Kost's fix for jpegenc. Fixes #591951.
+
+2009-08-14 20:18:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ 0.10.15.3 pre-release
+
+2009-08-14 13:45:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/elements/rtpbin.c:
+ checks: add test for leak to rtpbin unit test
+ See #591476.
+
+2009-08-11 14:47:12 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Fix reference leak
+ Fixes #591476.
+
+2009-08-14 13:34:53 +0100 Zaheer Merali <zaheerabbas@merali.org>
+
+ * ext/dv/gstdvdec.c:
+ dvdec: set bottom field first on PAL interlaced content, not top field first
+ DV interlaced content is always bottom field first. Fixes #591712.
+
+2009-08-14 12:44:06 +0100 Hans de Goede <jwrdegoede@fedoraproject.org>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: fix 'hang' with some cameras caused by bad timestamping if no framerate is available
+ For cameras/drivers that don't support e.g. VIDIOC_G_PARM we'd end up without
+ a framerate and would try to divide by 0, causing run-time warnings and all
+ frames to be timestamped with 0, which makes sinks that sync against the clock
+ drop them, causing 'hangs' (observed with the pwc driver and a Logitech QuickCam
+ Pro 4000). So if we do not know the framerate, simply don't adjust the
+ timestamps. Fixes #591451.
+
+2009-08-14 10:11:25 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: clear format list in READY->NULL
+ Clear format list and probed caps when going to NULL so if a new device
+ is set we'll probe the formats again instead of using previously
+ detected ones. Fixes bug #591747.
+
+2009-08-11 17:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ 0.10.15.2 pre-release
+
+2009-08-11 15:25:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * MAINTAINERS:
+ Add myself to MAINTAINERS file and update Wim's e-mail.
+
+2009-08-11 03:08:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/v4l2/Makefile.am:
+ v4l2: fix make distcheck by disting some more headers
+
+2009-08-11 02:42:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ docs: update
+
+2009-08-11 02:31:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/inspect/plugin-gstrtpmanager.xml:
+ * gst-plugins-good.spec.in:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/pipelines/.gitignore:
+ Move rtpmanager from -bad to -good.
+ Hook up build infrastructure (autotools, docs, unit test).
+
+2009-08-06 19:26:21 +0200 ric <csxnju at sogou.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: avoid buffer leak on bad seqnum
+ Fixes #590797
+
+2009-07-28 18:18:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ 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
+
+2009-07-28 11:59:56 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Incoming buffers do not always have caps
+
+2009-07-27 15:46:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ 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.
+
+2009-07-27 13:17:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ 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.
+
+2009-07-27 12:43:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ 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.
+
+2009-07-21 15:33:41 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Free conflicting addresses on finalize
+
+2009-07-01 12:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpbin: use new method for netaddress to string
+
+2009-06-29 18:48:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * tests/check/elements/rtpbin.c:
+ 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
+
+2009-05-28 19:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: add a comment
+
+2009-06-29 16:37:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpsession.c:
+ 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.
+
+2009-06-29 16:21:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ 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.
+
+2009-06-02 17:46:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/Makefile.am:
+ * gst/rtpmanager/gstrtpclient.c:
+ * gst/rtpmanager/gstrtpclient.h:
+ * gst/rtpmanager/gstrtpmanager.c:
+ rtpbin: removed old gstrtpclient
+
+2009-06-19 19:09:19 +0200 Branko Subasic <branko.subasic at axis.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ * tests/check/elements/rtpbin_buffer_list.c:
+ 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
+
+2009-06-19 16:21:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Make build without warnings with debugging disabled
+
+2009-05-28 17:37:44 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Transform the right session sdes message
+ Fixes #584165
+
+2009-05-28 17:33:10 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ Add ssrc to application/x-rtp-source-sdes structure
+
+2009-05-27 11:03:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsouce: the network address is in network order
+ Bring the network address in netowkr byte order to the host order.
+
+2009-05-26 15:40:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ 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.
+
+2009-05-25 13:46:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: remove ptdemux ghostpads
+
+2009-05-25 13:33:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/rtpbin.c:
+ tests: add receive rtpbin unit test
+
+2009-05-22 16:41:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: add to new signal to remove SSRC pads
+
+2009-05-22 16:35:20 +0200 Ali Sabil <ali.sabil at gmail.com>
+
+ * gst/rtpmanager/gstrtpbin-marshal.list:
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/rtpmanager/gstrtpssrcdemux.h:
+ ssrcdemux: emit signal when pads are removed
+ Add action signal to clear an SSRC in the ssrc demuxer.
+ Add signal to notify of removed ssrc.
+ See #554839
+
+2009-05-22 15:45:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ 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.
+
+2009-05-22 15:37:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/rtpbin.c:
+ tests: more rtpbin checks
+
+2009-05-22 15:36:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: don't warn when getting request pads twice
+ Allow getting the request pads multiple times, just return the previously
+ created pads.
+
+2009-05-22 13:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: add RTP and RTCP source address
+ Add the RTP and RTCP sender addresses in the stats structure.
+
+2009-05-22 13:45:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: reuse source code for SDES
+ Reuse the RTPSource object property instead of duplicating code.
+
+2009-05-22 13:44:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/rtpbin.c:
+ tests: add more rtpbin tests
+
+2009-05-22 12:23:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/rtpbin.c:
+ tests: add rtpbin unit test
+ Add the beginnings of an rtpbin unit test
+ Add some more stuff to .gitignore
+
+2009-05-22 12:20:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ 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
+
+2009-05-22 11:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: unref requests pads after releasing
+
+2009-05-22 01:43:50 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Implement releasing the streams
+ See #561752
+
+2009-05-22 01:16:11 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Keep jb signals handler
+ Keep the signal handlers so they can be disconnected at release time
+ See #561752
+
+2009-05-22 01:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: use the right lock for the sessions
+ Use the right lock when iterating the sessions.
+
+2009-05-22 01:03:55 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ 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
+
+2009-05-22 00:51:53 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Implement relasing of the rtp recv pad
+
+2009-05-22 00:44:51 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Implement releasing of rtp send pads
+
+2009-05-22 00:34:36 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Implement release of the recv rtcp pad
+ See #561752
+
+2009-05-22 00:16:19 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Implement releasing of rtcp src pad
+ See #561752
+
+2009-05-05 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ rtpssrcdemux: drop unexpected RTCP packets
+ We usually only get SR packets in our chain function but if an invalid packet
+ contains the SR packet after the RR packet, we must not fail but simply ignore
+ the malformed packet.
+ Fixes #581375
+
+2009-04-27 11:09:08 +0200 Olivier Crete <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ 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
+
+2009-04-27 11:06:01 +0200 Olivier Crete <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: notify when SSRC changes
+ Emit a g_object_notify when the SSRc changes because of a collision.
+ Fixes #580144
+
+2009-04-17 16:16:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: join the RTCP thread
+ Avoid a case where a joinable thread would be left unjoined, which leaked the
+ thread structure.
+ Fixes #577318.
+
+2009-04-15 18:14:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ 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.
+
+2009-04-15 17:44:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: we should not provide a clock
+ There is no need to provide a clock.
+
+2009-04-15 17:28:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: more estimated EOS fixes
+ Do more accurate EOS estimate and guard against backward timestamps.
+
+2009-04-15 17:25:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: release lock before pushing EOS
+ Make sure we release the jitterbuffer lock before we start pushing out data
+ because else we might deadlock.
+
+2009-03-27 17:44:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.h:
+ 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.
+
+2009-03-13 15:59:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpbin: don't return FALSE on seek events
+ Silently ignore the seek event instead of returning FALSE.
+
+2009-02-26 13:10:29 +0100 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ gstrtpbin: Don't forward revc events to sender
+ Don't send events from the receiver to the sender side.
+ Fixes #572900.
+
+2009-02-25 11:45:05 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ 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.
+
+2009-01-23 12:13:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpstats.c:
+ 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.
+
+2009-01-22 13:33:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ 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.
+
+2009-01-02 17:40:06 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2009-01-02 16:50:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-12-29 15:49:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-12-29 15:21:58 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-12-29 14:21:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-12-23 11:39:59 +0000 Arnout Vandecappelle <arnout@mind.be>
+
+ 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.
+
+2008-11-26 12:40:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-26 12:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-26 11:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-25 15:12:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-22 15:31:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-22 15:24:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-22 13:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-21 16:17:22 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2008-11-20 18:41:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-20 08:19:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ 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.
+
+2008-11-19 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-19 09:06:29 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-17 19:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ 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.
+
+2008-11-17 15:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-13 15:48:54 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-10 15:26:40 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-11-04 12:42:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ 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.
+
+2008-10-16 13:05:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-10-16 09:51:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-10-07 18:54:41 +0000 Håvard Graff <havard.graff@tandberg.com>
+
+ gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.
+ Original commit message from CVS:
+ Patch by: Håvard 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.
+
+2008-10-07 11:33:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-10-07 10:02:20 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-09-30 15:08:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-09-23 18:13:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-09-17 13:59:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ 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
+
+2008-09-13 01:37:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-09-05 13:52:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-08-28 15:21:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-08-13 14:31:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-08-11 07:20:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ 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?
+
+2008-08-05 09:42:53 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2008-08-05 09:00:50 +0000 Håvard Graff <havard.graff@tandberg.com>
+
+ gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum.
+ Original commit message from CVS:
+ Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_rtp_jitter_buffer_chain):
+ Fix debug by logging the right seqnum.
+
+2008-08-05 08:58:27 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2008-07-03 14:44:51 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ 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).
+
+2008-07-03 14:31:10 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ 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.
+
+2008-07-03 13:47:19 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ 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().
+
+2008-06-16 07:30:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ 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.
+
+2008-06-16 07:03:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ 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.
+
+2008-06-12 14:49:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ 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.
+
+2008-06-06 13:01:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-05-27 16:48:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-05-26 10:09:29 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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..
+
+2008-05-26 10:00:24 +0000 Håvard Graff <havard.graff@tandberg.com>
+
+ gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to the jitterbuffers when they are changed o...
+ Original commit message from CVS:
+ Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+ * gst/rtpmanager/gstrtpbin.c:
+ (gst_rtp_bin_propagate_property_to_jitterbuffer),
+ (gst_rtp_bin_set_property):
+ Propagate the do-lost and latency properties to the jitterbuffers when
+ they are changed on rtpbin.
+
+2008-05-26 09:57:40 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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().
+
+2008-05-16 19:56:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ 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
+
+2008-05-14 21:02:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-05-13 09:06:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-05-12 18:43:41 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-05-12 14:28:09 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ 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.
+
+2008-05-12 14:12:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ 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.
+
+2008-05-09 07:41:58 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ 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.
+
+2008-05-08 09:43:33 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2008-05-08 06:23:39 +0000 Sjoerd Simons <sjoerd@luon.net>
+
+ 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.
+
+2008-04-25 11:32:09 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-04-25 11:22:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-04-25 09:35:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-04-25 08:21:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-04-25 08:15:58 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-04-21 08:26:37 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2008-04-17 07:31:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-04-09 22:27:50 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ 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.
+
+2008-03-11 12:40:58 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2008-03-11 11:36:03 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+ gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked d...
+ Original commit message from CVS:
+ Based on patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
+ (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread),
+ (join_rtcp_thread), (gst_rtp_session_change_state):
+ Avoid a deadlock when joining the RTCP thread in PAUSED because it might
+ be blocked downstream. Also avoid spawning multiple rtcp threads.
+ Fixes #520894.
+
+2008-03-11 10:43:32 +0000 Stefan Kost <ensonic@users.sf.net>
+
+ gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we have no timestamps.
+ Original commit message from CVS:
+ Patch by: Stefan Kost <ensonic@users.sf.net>
+ * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
+ Don't try to reset the clock skew when we have no timestamps.
+ Fixes #519005.
+
+2008-02-20 09:33:25 +0000 Olivier Crete <tester@tester.ca>
+
+ 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.
+
+2008-02-14 16:25:51 +0000 Olivier Crete <tester@tester.ca>
+
+ gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160.
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester@tester.ca>
+ * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate):
+ Ignore streams that did not receive an SR packet when doing
+ synchronisation. Fixes #516160.
+
+2008-01-29 18:57:27 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ 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.
+
+2008-01-25 16:58:00 +0000 Olivier Crete <tester@tester.ca>
+
+ gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not provided with caps but with a signal.
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester@tester.ca>
+ * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
+ (create_stream), (payload_type_change), (new_ssrc_pad_found):
+ Also handle lip-sync when the clock-rate is not provided with caps but
+ with a signal.
+
+2008-01-25 16:00:52 +0000 Olivier Crete <tester@tester.ca>
+
+ gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided...
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester@tester.ca>
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
+ * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
+ (rtp_jitter_buffer_insert):
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ Remove the fixed clock-rate from the jitterbuffer and extend it so that
+ a clock-rate can be provided with each buffer instead. Fixes #511686.
+
+2008-01-25 15:49:55 +0000 Olivier Crete <tester@tester.ca>
+
+ gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester@tester.ca>
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state),
+ (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+ Remove old unused variable.
+ Track pt on input buffers and get the clock-rate when it changes.
+ Ignore packets with unknown clock-rate. See #511686.
+
+2008-01-25 01:44:27 +0000 Olivier Crete <tester@tester.ca>
+
+ gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. Fixes #511920
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester@tester.ca>
+ * gst/rtpmanager/rtpsource.c: Fix unref of buffer using the
+ wrong function. Fixes #511920
+
+2008-01-11 17:02:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-01-11 16:45:57 +0000 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ 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.
+
+2008-01-10 20:57:17 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
+ Original commit message from CVS:
+ * gst/rtpmanager/gstrtpbin.c:
+ Fix documentation for latest patch
+
+2008-01-10 14:34:30 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ 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)
+
+2008-01-09 14:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-01-04 18:47:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-12-31 13:12:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-12-12 16:59:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-12-12 12:11:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-12-10 18:36:04 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-12-10 15:34:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-12-10 14:03:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-12-10 11:08:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-11-22 09:08:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-11-02 21:45:38 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+ gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst/rtpmanager/gstrtpsession.c:
+ Fix bad function signatures (#492798).
+
+2007-10-09 10:01:39 +0000 Laurent Glayal <spglegle@yahoo.fr>
+
+ 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.
+
+2007-10-08 17:46:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ 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.
+
+2007-10-08 10:39:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-10-05 17:26:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-10-05 12:07:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-10-02 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-28 14:51:58 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-28 11:17:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-26 20:08:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-20 14:34:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-17 02:01:41 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-16 19:40:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-15 18:48:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-12 21:23:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-12 18:04:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-09-04 15:23:34 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ 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).
+
+2007-09-03 21:19:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-31 15:26:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-29 16:56:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-29 01:22:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-28 20:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-27 21:17:21 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-23 21:39:58 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ 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.
+
+2007-08-21 17:18:29 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-21 16:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-16 11:40:16 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-08-13 06:16:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/rtpmanager/rtpjitterbuffer.c: Include stdlib.
+ Original commit message from CVS:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ Include stdlib.
+
+2007-08-10 17:16:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ 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).
+
+2007-06-22 20:23:18 +0000 Jens Granseuer <jensgr@gmx.net>
+
+ 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.
+
+2007-05-28 16:37:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-05-23 13:08:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-05-15 13:29:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-05-15 03:45:45 +0000 David Schleef <ds@schleef.org>
+
+ 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.
+
+2007-05-14 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-05-10 14:02:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
+ Original commit message from CVS:
+ * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
+ gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
+ gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
+ gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
+ qtdemux_parse_segments, qtdemux_parse_trak):
+ * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
+ rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
+ rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
+ rtp_session_get_location, rtp_session_get_tool,
+ rtp_session_process_bye, session_report_blocks):
+ * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
+ rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
+ More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
+ * gst/switch/Makefile.am:
+ Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
+
+2007-05-10 12:38:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ * gst/rtpmanager/async_jitter_queue.c:
+ gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a...
+ Original commit message from CVS:
+ * gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
+ async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
+ async_jitter_queue_set_low_threshold,
+ async_jitter_queue_length_ts_units_unlocked,
+ async_jitter_queue_unref_and_unlock, async_jitter_queue_unref,
+ async_jitter_queue_lock, async_jitter_queue_push,
+ async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted,
+ async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop,
+ async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
+ async_jitter_queue_set_flushing_unlocked,
+ async_jitter_queue_unset_flushing_unlocked):
+ Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
+
+2007-05-09 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.
+ Original commit message from CVS:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_rtp_jitter_buffer_query):
+ Pass queries upstream.
+
+2007-05-04 12:32:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-30 13:41:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-29 14:46:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-27 15:09:12 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-25 16:38:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-25 15:48:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-25 13:19:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-25 08:30:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-21 19:21:49 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ 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.
+
+2007-04-18 18:58:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-13 09:20:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-12 08:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-11 13:49:54 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-10 09:14:07 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-06 12:28:29 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-06 12:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-05 16:10:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-05 13:54:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-04 10:23:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-03 11:35:39 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-03 09:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2007-04-03 09:13:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add RTP session management elements. Still in progress.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/rtpmanager/Makefile.am:
+ * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
+ (signal_waiting_threads), (async_jitter_queue_ref),
+ (async_jitter_queue_ref_unlocked),
+ (async_jitter_queue_set_low_threshold),
+ (async_jitter_queue_set_high_threshold),
+ (async_jitter_queue_set_max_queue_length),
+ (async_jitter_queue_get_g_queue), (calculate_ts_diff),
+ (async_jitter_queue_length_ts_units_unlocked),
+ (async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
+ (async_jitter_queue_lock), (async_jitter_queue_unlock),
+ (async_jitter_queue_push), (async_jitter_queue_push_unlocked),
+ (async_jitter_queue_push_sorted),
+ (async_jitter_queue_push_sorted_unlocked),
+ (async_jitter_queue_insert_after_unlocked),
+ (async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
+ (async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
+ (async_jitter_queue_length_unlocked),
+ (async_jitter_queue_set_flushing_unlocked),
+ (async_jitter_queue_unset_flushing_unlocked),
+ (async_jitter_queue_set_blocking_unlocked):
+ * gst/rtpmanager/async_jitter_queue.h:
+ * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
+ (gst_rtp_bin_class_init), (gst_rtp_bin_init),
+ (gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
+ (gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
+ (gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
+ (free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
+ (gst_rtp_client_class_init), (gst_rtp_client_init),
+ (gst_rtp_client_finalize), (gst_rtp_client_set_property),
+ (gst_rtp_client_get_property), (gst_rtp_client_change_state),
+ (gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
+ * gst/rtpmanager/gstrtpclient.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_rtp_jitter_buffer_base_init),
+ (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
+ (gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
+ (gst_jitter_buffer_sink_setcaps), (free_func),
+ (gst_rtp_jitter_buffer_flush_start),
+ (gst_rtp_jitter_buffer_flush_stop),
+ (gst_rtp_jitter_buffer_src_activate_push),
+ (gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
+ (compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
+ (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
+ (gst_rtp_jitter_buffer_query),
+ (gst_rtp_jitter_buffer_set_property),
+ (gst_rtp_jitter_buffer_get_property):
+ * gst/rtpmanager/gstrtpjitterbuffer.h:
+ * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
+ * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
+ (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
+ (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
+ (gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
+ (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
+ (gst_rtp_pt_demux_change_state):
+ * gst/rtpmanager/gstrtpptdemux.h:
+ * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
+ (gst_rtp_session_class_init), (gst_rtp_session_init),
+ (gst_rtp_session_finalize), (gst_rtp_session_set_property),
+ (gst_rtp_session_get_property), (gst_rtp_session_change_state),
+ (gst_rtp_session_chain_recv_rtp),
+ (gst_rtp_session_chain_recv_rtcp),
+ (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
+ (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
+ (gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
+ * gst/rtpmanager/gstrtpsession.h:
+ Add RTP session management elements. Still in progress.
+
+2009-08-10 13:30:23 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: push mode; cater for chunk padding
+
+2009-08-04 19:45:43 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: only use stream's pad after having checked it exists
+
+2009-08-04 13:38:09 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: sprinkle some more GST_DEBUG_FUNCPTR
+
+2009-08-04 13:36:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: post error message if no pads to push EOS event on
+
+2009-08-04 11:39:59 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix typo in warning message
+
+2009-08-04 11:39:39 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix some buffer ref handling
+
+2009-08-04 11:37:16 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: do not exceed maximum number of supported streams
+
+2009-08-04 11:35:18 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: prevent double unref; gst_avi_demux_parse_avih already unrefs
+
+2009-08-04 11:32:27 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: verify size of INFO LIST to satisfy subsequent expectations
+
+2009-07-29 15:25:38 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: check video stream framerate against avi header frame duration
+ The former might be bogus in silly cases, and the latter seems to
+ carry more weight.
+
+2009-08-04 12:16:13 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: streamline stream duration calculation
+
+2009-07-03 14:04:13 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/raw1394/gstdv1394src.c:
+ dv1394src: Fix element for live usage... which has been broken for 2 years :(
+ This is a live source, therefore:
+ * Use GST_FORMAT_TIME as the default format
+ * set_timestamp to True
+ * properly implement query latency.
+ This allows expected live usage like : playbin2 uri=dv://
+
+2009-08-09 09:43:41 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/raw1394/gstdv1394src.c:
+ raw1394: Remove unneeded variable
+
+2009-08-09 09:43:29 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska: remove dead assignments
+
+2009-08-09 09:43:00 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ rtp: Remove dead assignments and resulting unneeded variables.
+
+2009-08-10 09:53:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ * ext/wavpack/Makefile.am:
+ * ext/wavpack/gstwavpackenc.c:
+ * ext/wavpack/gstwavpackenc.h:
+ * ext/wavpack/md5.c:
+ * ext/wavpack/md5.h:
+ wavpack: Use GLib GChecksum instead of our own MD5 implementation
+ This requires GLib 2.16 but that version is already required by core anyway.
+
+2009-08-08 00:47:48 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroska: Adds support to muxing/demuxing WMA
+ Adds support for muxing wma audio family and fixes
+ demuxing of wma family in matroskademux. matroskademux
+ was broken because it missed codec_data.
+
+2009-08-06 20:15:17 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: adds support for wmv family
+ Adds support to WMV1, WMV2, WMV3 and other family formats that
+ are signaled by the 'format' field in the caps (i.e. WVC1).
+ Partially fixes #576378
+
+2009-08-09 14:19:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2src: if max == min width/height put an int in the probed caps, not an int range
+ Fixes #560033.
+
+2009-08-09 13:58:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudiosrc: if max_channels == min_channels, use an int instead of an int range in the caps
+
+2009-08-09 12:52:17 +0200 LoneStar <lone@auvtech.com>
+
+ * gst/id3demux/id3v2frames.c:
+ id3demux: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8
+ Fixes bug #499242.
+
+2009-08-09 01:29:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to latest release
+ To avoid confusion.
+
+2009-08-09 01:27:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/elements/flvmux.c:
+ check: fix flvmux unit test on big endian machines
+ flvmux only accepts raw audio in little endian, but audiotestsrc
+ produces audio in the native endianness, which makes linking
+ between audiotestsrc and flvmux fail on big endian machines. Add
+ an audioconvert element in between the two to fix this.
+
+2009-02-15 18:49:44 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: add kate subtitle support to matroska muxer and demuxer
+ See #525743.
+
+2009-08-07 16:51:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/id3demux/id3v2.3.0.html:
+ id3demux: add ID3 v2.3 spec as well
+
+2009-08-07 16:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/id3demux/id3v2frames.c:
+ id3demux: sizes in ID3 v2.3 are unlikely to be sync-safe integers
+ In ID3 v2.3 compressed frames will have a 4-byte data length indicator
+ after the frame header to indicate the size of the decompressed data.
+ This integer is unlikely to be a sync-safe integer for v2.3 tags,
+ only in v2.4 it's sync-safe.
+
+2009-08-07 16:36:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/id3demux/id3tags.c:
+ id3demux: fix typo in debug message
+
+2009-08-07 16:02:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/id3demux/id3tags.c:
+ * gst/id3demux/id3tags.h:
+ * gst/id3demux/id3v2frames.c:
+ * tests/check/elements/id3demux.c:
+ * tests/files/Makefile.am:
+ * tests/files/id3-588148-unsynced-v24.tag:
+ id3demux: fix parsing of unsync'ed ID3 v2.4 tags and frames
+ Reversing the unsynchronisation seems to work slightly differently
+ for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame
+ sizes in the frame header, so the unsynchronisation is applied to
+ the whole frame data including all the frame headers. v2.4 frames
+ have sync-safe sizes, however, so the unsynchronisation only needs
+ to be applied to the actual frame data, and it seems that's what's
+ being done as well. So we need to undo the unsynchronisation on a
+ per-frame basis for v2.4 tags for things to work properly.
+ Fixes extraction of coverart/images from APIC frames in ID3 v2.4
+ tags (#588148).
+ Add unit test for this as well.
+
+2009-08-06 21:24:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Use SOUP_METHOD_GET instead of "GET" string
+ Fixes bug #590970.
+
+2009-08-06 13:00:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: set the default slave method to skew
+ Set the default slave method to the much better skew algorithm. This is the
+ default in the new base class but we override this here as well for the
+ upcomming release.
+
+2009-08-06 10:20:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: fix compilation with --disable-gst-debug
+
+2009-08-03 18:59:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ rtph264pay: use array instead of queue
+
+2009-08-03 18:55:19 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ rtph264pay: push NALs only after SPS/PPS
+ parse complete (bytestream) buffer for SPS/PPS before pushing NALs.
+ Fixes #564501.
+
+2009-08-04 14:44:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * sys/v4l2/v4l2_calls.h:
+ v4l2: Directly use GST_PTR_FORMAT for printing caps with the LOG_CAPS macro
+
+2009-08-04 11:17:17 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpqdmdepay.c:
+ rtpqdm2depay: Fix debug statement.
+
+2009-08-04 09:32:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/v4l2_calls.h:
+ v4l2: Remove some OMAP specific hacks
+ They require special build flags and are not useful in general.
+
+2009-08-04 09:22:29 +0200 Rob Clark <rob@ti.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/v4l2src_calls.c:
+ v4l2sink: change where buffers get dequeued
+ It seems to cause strange occasional high latencies (almost 200ms) when dequeuing buffers from _buffer_alloc(). It is simpler and seems to work much better to dqbuf from the same thread that is queuing the next buffer.
+
+2009-08-04 09:14:20 +0200 Rob Clark <rob@ti.com>
+
+ * sys/v4l2/Makefile.am:
+ * sys/v4l2/gstv4l2.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/gstv4l2sink.h:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ * sys/v4l2/v4l2_calls.c:
+ * sys/v4l2/v4l2_calls.h:
+ * sys/v4l2/v4l2src_calls.c:
+ * sys/v4l2/v4l2src_calls.h:
+ v4l2: Add v4l2sink element
+ This also does the following changes:
+ (1) pull the bufferpool code out into gstv4l2bufferpool.c, and make a
+ bit more generic so it can be used both for v4l2src and v4l2sink
+ (2) move some of the device probing/configuration/caps stuff into
+ gstv4l2object.c so it does not have to be duplicated between
+ v4l2src and v4l2sink
+ Fixes bug #590280.
+
+2009-08-04 07:07:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ flvmux: Enable unit test now that it passes
+
+2009-08-03 21:21:39 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ rtpqdm2depay,rtpsv3vdepay: Add debugging category.
+
+2009-08-03 21:22:48 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpqdmdepay.h:
+ rtpqdm2depay: Handle gaps in incoming packets.
+ Whenever we see a gap, we flush the temporary packets (but not the adapter). If we
+ had some data temporarily stored it will be outputted (the sound will sound a bit
+ garbled... but that's how it sounds on MacOSX :)
+
+2009-08-03 19:01:07 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpqdmdepay.c:
+ rtpqdmdepay: Fix CRC calculation and remove commented code.
+
+2009-08-02 13:42:12 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpqdmdepay.h:
+ rtp: New QDM2 rtp depayloader.
+ Reverse-engineered by comparing:
+ * A rtp hinted file provided by DarwinStreamingServer
+ * The output procued by DSS for that same file
+ Also used various streaming sources available on the internet to fine-tune
+ the code.
+ The header/codec_data extraction methods are from FFMpeg (LGPL).
+
+2009-08-03 21:24:44 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpsv3vdepay.c:
+ rtpsv3vdepay: Properly fill codec_data and cleanup code a bite more.
+
+2009-08-03 19:02:17 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtpsv3vdepay.h:
+ rtpsv3vdepay: Only output buffers once we're configured.
+
+2009-08-03 19:02:00 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpsv3vdepay.c:
+ rtpsv3vdepay: Add more encoding-name variants
+
+2009-08-03 20:08:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/elements/flvmux.c:
+ flvmux: Fix unit test to correctly handle request pads
+ Request pads are removed by the element instance in PAUSED->READY
+ so we need to re-request pads for every run and link them again.
+ Last fix for bug #590447.
+
+2009-08-03 20:08:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Fix writing of the index for < 128 buffers
+ Partially fixes bug #590447.
+
+2009-08-03 20:07:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Fix resetting of the element
+ Reset the have_video/have_audio flags and make sure to
+ properly release the request pads.
+ Partially fixes bug #590447.
+
+2009-08-03 18:13:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: don't add non-utf8 chars to structures
+
+2009-08-03 18:02:31 +0200 Luc Deschenaux <luc.deschenaux at freesurf.ch>
+
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegdepay.h:
+ jpegdepay: use attributes for extra properties
+ Use some of the SDP attributes when they are present to specify the output
+ dimension and framerate. This allows us to receive jpeg frames larger than
+ 2040 width/height.
+ Fixes #564437
+
+2009-08-03 18:01:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/README:
+ RTP docs: update with attributes in caps
+
+2009-08-03 17:21:44 +0200 Luc Deschenaux <luc.deschenaux at freesurf.ch>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: put all SDP attributes on caps
+ Put the SDP attributes on the caps too so that they can be used by
+ depayloaders.
+ See #564437
+
+2009-08-03 13:32:12 +0200 Jonathan Tellier <jonathan.tellier at gmail.com>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: initialize the probe with the server
+ When creating a new probe, pass the server instead of the device string.
+ fixes #590401
+
+2009-08-02 11:44:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: don't do things with side-effects inside g_return_val_if_fail()
+ Someone might compile this code with -DG_DISABLE_ASSERT some day.
+
+2009-08-01 21:39:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: don't do logic within g_assert() statements
+ Otherwise that code will just be expanded to nothing when compiled
+ -DG_DISABLE_ASSERT (PS: why is mainloop_start() called in the init
+ function and not when changing state to READY?)
+
+2009-08-01 17:07:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: send newsegment event when operating push-based and unframed
+ For some reason flac doesn't call our metadata callback when we operate
+ in push mode with unframed input, but that's where we set up the
+ newsegment event (since that's where we'd get the duration from the
+ stream info header), so we didn't send a newsegment event at all in this
+ case. Hack around this by storing a generic newsegment event for now
+ which will be used if we don't replace it with a better one that
+ includes the duration.
+
+2009-08-01 16:48:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: small cleanups
+ Remove some callback indirections which are no longer needed because
+ there's only one decoder object type now. Also remove unused variable.
+
+2009-08-01 15:22:49 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: use gst_adapter_copy() to avoid unnecessary buffer merges
+ gst_adapter_peek() will merge buffers as needed, which we can avoid
+ here since we're doing a memcpy anyway and then flush the copied
+ data from the adapter right away.
+
+2009-08-01 00:00:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: repair some broken indenting
+
+2009-08-01 12:19:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/flvmux.c:
+ checks: add basic unit test for flvmux, but disable it for now
+ Basic unit test for flvmux. Fails miserably, hence disabled for now.
+
+2009-07-31 23:28:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/flvdemux.c:
+ * tests/files/Makefile.am:
+ * tests/files/pcm16sine.flv:
+ check: add basic unit test for flvdemux
+ In particular, test re-use of flvdemux in both pull and push mode
+ (see #583030).
+
+2009-07-31 20:25:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: fix invalid write caused by using sizeof("string") as length
+ sizeof("foo") includes the string's NUL-terminator in the size returned,
+ but we're writing strings here with an explicit size at the beginning
+ and no NUL-terminator. In most cases using sizeof("foo") as length in
+ memcpy is not harmful, but it is where the string goes right at the
+ end of our buffer to write, since we don't allocate space for that
+ NUL terminator.
+
+2009-07-27 18:44:45 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ soup: Use "GET" instead of SOUP_METHOD_GET. Fixes build with libsoup-2.7.*
+ This is due to a quality API change in libsoup 2.7. SOUP_METHOD_* are now
+ integers and not strings... they could have changed the names.
+
+2009-07-30 17:57:53 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/jpeg/gstjpegdec.c:
+ * ext/jpeg/gstjpegenc.c:
+ jpeg: use longer macro names to not clash with some stupid windows defines
+ libjpeg headers pull some windows system inlcudes (on windows) that contain a
+ define for DEFAULT_QUALITY.
+
+2009-07-29 14:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix last commit and improve readability
+
+2009-07-24 19:04:31 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * gst/avi/gstavidemux.c:
+ Fixed the fix for TIME->DEFAULT conversion.
+ Fixes bug #578052 again.
+
+2009-07-29 13:38:03 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtpsv3vdepay.c:
+ rtpsv3depay: Fix width/height calculation, bring up to marginal rank.
+ Based on documentation found on http://wiki.multimedia.cx/
+
+2009-07-29 12:13:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesrc.c:
+ pulse: conditionally compile newer stuff
+ configured_sink/source_usec in the timing_info is only since 0.9.11 so
+ conditionally compile this information.
+ fixes #590038
+
+2009-07-28 18:29:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesrc.c:
+ * ext/pulse/pulsesrc.h:
+ pulsesrc: cleanups
+ Keep track of the paused state of the source and leave the read function when
+ paused.
+ don't wait for a latency update when the delay is not yet known but simply
+ return 0 instead of blocking.
+ Keep track of the corked state of the stream.
+ Fix the state changes.
+
+2009-07-28 16:11:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: set maxlength always to -1
+
+2009-07-28 15:53:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesrc.c:
+ * ext/pulse/pulsesrc.h:
+ pulsesrc; cleanups, report real latency
+ Add some more debug info
+ Avoid some type casts
+ Report the real latency to the application.
+
+2009-07-28 16:11:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: when scanning for 0xff marker ends, ensure desired result
+ Otherwise, any non 0xff byte at end of data would be mistaken for
+ a tag byte, and in case of a frame_len 0 tag subsequently lead to an
+ infinite loop.
+
+2009-07-28 00:30:43 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+ * gst/avi/gstavimux.c:
+ avimux: adds support to wma
+
+2009-07-28 00:07:15 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+ * gst/avi/gstavimux.c:
+ avimux: adds support to wmv
+
+2009-07-27 21:34:22 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Downgrade warning message to debug
+
+2009-07-27 11:51:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: avoid using ivalid stream indexes
+ when we get an invalid stream index from pulse because we were just starting,
+ avoid using it for getting and setting the volume.
+ Fixes #589365
+
+2009-07-24 19:38:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstradioac.c:
+ * gst/effectv/gstripple.c:
+ * gst/effectv/gstshagadelic.c:
+ * gst/effectv/gststreak.c:
+ * gst/effectv/gstvertigo.c:
+ * gst/effectv/gstwarp.c:
+ effectv: Don't allow caps changes for some effectv filters
+ These filters use information from previous frames to
+ generate the current frame and a caps change will make
+ the effect start from the beginning again.
+
+2009-07-24 19:37:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstwarp.c:
+ * gst/effectv/gstwarp.h:
+ warptv: Make the sine table global instead of having it in every instance
+
+2009-07-24 10:47:44 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/jpeg/gstjpegenc.c:
+ jpeg: make encoder work with libjpeg v7
+ We have to specify do_fancy_downsampling = FALSE in the encoder with did not exist before.
+
+2009-07-24 00:42:33 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From fedaaee to 94f95e3
+
+2009-07-23 12:06:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: Implement SEEKING query
+ Fixes bug #589423.
+
+2009-07-22 11:16:06 +0100 Colin Guthrie <cguthrie@mandriva.org>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Fix a couple error messages that mentioned incorrect function names.
+ Fixes #589459.
+
+2009-07-23 11:50:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvparse.c:
+ flvdemux: Implement SEEKING query
+ Also add some more query types to the answer of the query type function.
+ Fixes bug #589424.
+
+2009-07-21 19:46:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ * ext/flac/gstflacdec.h:
+ flacdec: fix intermittent FLAC__STREAM_DECODER_ABORTED errors when seeking
+ When seeking in a local flac file (ie. operating pull-based), the decoder
+ would often just error out after the loop function sees a DECODER_ABORTED
+ status. This, however, is the read callback's way of telling our loop
+ function that pull_range failed and streaming should stop, in this case
+ because of the flush-start event that the seek handler pushed upstream
+ from the seeking thread. Handle this slightly better by storing the last
+ flow return from pull_range, so the loop function can evaluate it properly
+ when it encounters a DECODER_ABORTED and take the right action.
+ Fixes #578612.
+
+2009-07-21 10:07:00 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/interleave/interleave.c:
+ interleave: fix indenting and upgrade two debugs to warnings.
+ Fix newlines in variable decls. Change two debugs to become warnings as they
+ indicate that things will not work.
+
+2009-07-21 10:04:36 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/jpeg/gstjpeg.c:
+ * ext/jpeg/gstjpegdec.c:
+ * ext/jpeg/gstjpegenc.c:
+ * ext/jpeg/gstjpegenc.h:
+ jpeg: code cleanups for encoder
+ Remove some disabled code in encoder. Try #if 0'ed code and add comments about
+ why it is disabled. Move idct-method enum to jpeg.c and use in both encoder and
+ decoder. Add idct-method property to encoder.
+
+2009-07-21 07:50:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Answer SEEKING queries in the original format
+
+2009-07-21 01:12:44 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * gst/udp/gstudpnetutils.c:
+ udputils: initialize struct content with 0.
+ Fixes some random crashes.
+
+2009-07-20 19:09:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: set some values to their defaults
+ Set the minreq and maxlength buffer attributes to -1 to let puleseaudio select a
+ sensible value.
+
+2009-07-20 19:04:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: don't wait for posted message
+ We can't wait for the ENTER/LEAVE messages to be be posted because the base
+ class sometimes calls the start method with the object lock, which would block
+ the message posting.
+ Instead, just assume that the message will be posted soon and continue. We'll
+ have to fix this in the base class.
+
+2009-07-20 18:11:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: use relative seeks
+ Use relative seeks because I was told that absolute seeks don't work.
+
+2009-07-20 16:52:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Implement SEEKING query
+
+2009-07-20 08:07:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ cairorender: Add support for ARGB/BGRA input
+ Note that videotestsrc outputs 100% transparent video
+ which will result in white output from cairorender.
+
+2009-07-17 13:22:57 +0100 Elaine Xiong <Elaine.Xiong@Sun.COM>
+
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/v4l2_calls.c:
+ * sys/v4l2/v4l2src_calls.c:
+ v4l2: Fix v4l2src on OpenSolaris
+ The v4l2 driver for USB webcams on OpenSolaris does not support select()
+ calls. Detect when select() fails, and skip polling the device afterward,
+ which restores the pre 0.10.14 behaviour on OpenSolaris.
+ Signed-off-by: Jan Schmidt <thaytan@noraisin.net>
+
+2009-07-17 11:22:06 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * tests/check/elements/.gitignore:
+ * tests/examples/v4l2/.gitignore:
+ gitignore: Ignore some new binaries
+
+2009-07-17 13:49:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * ext/cairo/gstcairorender.c:
+ cairorender: Add to the documentation
+
+2009-07-17 13:42:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ cairorender: Return not-negotiated if we have no caps
+
+2009-07-17 13:41:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ * ext/cairo/gstcairorender.h:
+ cairorender: Fix caps and colorspace handling
+
+2009-07-17 13:30:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ cairorender: Use correct mimetypes for PDF and SVG
+
+2009-07-17 13:24:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ cairorender: Remove pull mode, it only adds complexity but not advantages
+
+2009-07-16 21:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ cairorender: Fix caps negotiation and cairo surface creation
+
+2009-07-16 21:42:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ cairorender: Correctly set srccaps
+
+2009-07-16 21:31:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ * ext/cairo/gstcairorender.h:
+ cairorender: Move instance/class struct definitions to the header
+
+2009-07-16 21:30:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/cairo/gstcairorender.c:
+ * ext/cairo/gstcairorender.h:
+ cairorender: Add Lutz' copyright to the file header
+
+2009-07-16 21:27:45 +0200 Lutz Mueller <lutz@topfrose.de>
+
+ * ext/cairo/Makefile.am:
+ * ext/cairo/gstcairo.c:
+ * ext/cairo/gstcairorender.c:
+ * ext/cairo/gstcairorender.h:
+ cairo: Add cairo-based PDF/PS/SVG encoder element
+ Fixes bug #331420.
+
+2009-07-16 20:44:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/flac/gstflacenc.c:
+ * ext/flac/gstflacenc.h:
+ flacenc: Optionally write a PADDING block
+ The size of the PADDING block is specified by a new
+ "padding" property.
+ Fixes bug #588483.
+
+2009-07-16 19:35:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Only assume seekability if the server provides Content-Length
+ Previously seekability way always assumed until the first seek actually
+ failed. Now we assume that all servers are not seekable unless they provide
+ a Content-Length header. If a seek fails after that we continue to
+ assume no seekability. Fixes bug #585576.
+
+2009-07-16 15:14:43 +0200 Arnout Vandecappelle <arnout@mind.be>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: don't try to authenticate if no username/password is set.
+
+2009-07-16 17:10:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstwarp.c:
+ effectv: Chain up finalize to the parent class in warptv
+ Fixes a memory leak.
+
+2009-07-16 12:55:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/effectv.c:
+ effectv: Add unit test for all effectv elements
+
+2009-07-16 12:17:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ effectv: Add new effectv elements to the docs
+
+2009-07-15 14:37:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/Makefile.am:
+ * gst/effectv/gsteffectv.c:
+ * gst/effectv/gstripple.c:
+ * gst/effectv/gstripple.h:
+ effectv: Add rippletv element
+ This produces a water ripple effect on the video input,
+ based on motion or a rain drop algorithm.
+ Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+ Fixes bug #588695.
+
+2009-07-12 15:42:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/Makefile.am:
+ * gst/effectv/gsteffectv.c:
+ * gst/effectv/gststreak.c:
+ * gst/effectv/gststreak.h:
+ effectv: Add streaktv effect filter element
+ This combines the StreakTV and BaltanTV filters from the
+ effectv project.
+ Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+ Fixes bug #588368.
+
+2009-07-12 12:31:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstedge.c:
+ * gst/effectv/gstop.c:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstradioac.c:
+ * gst/effectv/gstrev.c:
+ * gst/effectv/gstshagadelic.c:
+ * gst/effectv/gstvertigo.c:
+ effectv: Fix processing on big endian architectures
+
+2009-07-12 11:52:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/Makefile.am:
+ * gst/effectv/gsteffectv.c:
+ * gst/effectv/gstradioac.c:
+ * gst/effectv/gstradioac.h:
+ effectv: Add radioactv effect filter
+ This filter adds a radiation-like motion blur effect
+ to the video stream.
+ Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+ Fixes bug #588359.
+
+2009-07-12 11:26:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstop.c:
+ * gst/effectv/gstop.h:
+ effectv: Make the optv threshold property an uint
+
+2009-07-12 10:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/Makefile.am:
+ * gst/effectv/gsteffectv.c:
+ * gst/effectv/gstop.c:
+ * gst/effectv/gstop.h:
+ effect: Add optv effect filter from the effectv project
+ This filter binarizes input frames and combines them with various
+ optical pattern.
+ Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+ Fixes bug #588349.
+
+2009-07-03 05:11:26 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Emit stream-status leave message
+ Fixes #587695
+
+2009-07-03 05:06:45 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesink.h:
+ pulsesink: Emit stream-status enter message
+ Emit stream-status messages for the pulse thread.
+ Don't use our own GCond for signaling but simply use the pulse mainloop
+ mechanisms for synchronisation.
+ See #587695
+
+2009-07-14 18:15:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: debug the latency update values
+
+2009-07-14 16:12:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulseutil.c:
+ pulsesink: add 24bit sample formats
+ Add check for pulseaudio 0.9.15 and enable 24bits samples in that case.
+
+2009-07-13 12:23:37 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 5845b63 to fedaaee
+
+2009-07-13 17:53:25 +0200 Marc Leeman <marc.leeman at gmail.com>
+
+ * gst/rtp/gstrtpmpvpay.c:
+ mpvpay: Rework the timestamping
+ Rework the timestamping in the mpv payloader so that the timestamps are more
+ accurate.
+ Fixes #587680
+
+2009-07-03 08:47:12 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
+
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/v4l2/Makefile.am:
+ * tests/examples/v4l2/probe.c:
+ v4l2src: add a simple test case for device probing
+
+2009-07-03 08:38:43 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
+
+ * configure.ac:
+ * sys/v4l2/Makefile.am:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2src: optional support for device probing with gudev
+ Enumerate v4l2 devices using gudev if available.
+ Fixes bug #583640.
+
+2009-07-10 19:54:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Random cleanup
+
+2009-07-10 19:54:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Send queries to the master pad by default instead of all pads
+
+2009-07-10 19:34:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/blend_rgb.c:
+ * gst/videomixer/videomixer.c:
+ videomixer: Add RGB, BGR, xRGB, RGBx, xBGR, BGRx support
+
+2009-07-10 17:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Clean up debugging a bit
+
+2009-07-10 17:25:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Remove some redundant checks and error out immediately if not negotiated
+ Also stop leaking the output buffer in some error cases.
+
+2009-07-10 17:23:03 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/blend_i420.c:
+ * gst/videomixer/videomixer.c:
+ * gst/videomixer/videomixer.h:
+ videomixer: Remove the calculate_frame_size() function and use libgstvideo instead
+
+2009-06-30 15:13:44 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Remove unused link/unlink pad methods
+
+2009-06-30 12:43:04 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/videomixer/blend_i420.c:
+ videomixer: I420 mode: Add fast path for 0.0 and 1.0 alpha
+ If the source alpha is 0.0, we take nothing.
+ If the source alpha is 1.0, we overwrite everything.
+
+2009-06-30 12:40:02 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/videomixer/blend_i420.c:
+ videomixer: I420 blending : Fix main algorithm.
+ When blending a source layer with an alpha of 'a' on top of another
+ destination layer we take the sum of:
+ * 'a' percent of the source layer
+ * (100 - 'a') percent of the destination layer (the remainder)
+
+2009-06-30 12:39:19 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/videomixer/blend_i420.c:
+ * gst/videomixer/videomixer.c:
+ * gst/videomixer/videomixer.h:
+ * gst/videomixer/videomixerpad.h:
+ videomixer: Make debugging category global to all the code.
+
+2009-06-29 19:23:41 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: improve readability of debugging statements.
+
+2009-07-08 13:38:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: do not leak timeout message
+
+2009-07-09 07:14:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avi: Don't forward NEWSEGMENT events from upstream
+ New ones are generated later and simply forwarding them can
+ result in NEWSEGMENT events of different format going downstream.
+ Fixes bug #587983.
+
+2009-07-08 18:19:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/blend_i420.c:
+ videomixer: Make checker pattern lookup table constant
+
+2009-07-08 18:17:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/videomixer.c:
+ videomixer: Add support for ARGB
+ And clean up the caps parsing.
+
+2009-07-08 15:17:41 +0200 Benjamin Gaignard <benjamin@gaignard.net>
+
+ * gst/udp/gstudpnetutils.c:
+ udp: Initialize pointer to NULL
+ Otherwise we're calling free() with some random
+ memory address in error cases.
+ Fixes bug #587982.
+
+2009-07-07 16:35:24 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: sprinkle some more const
+
+2009-07-07 15:57:55 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: perform some more (careful) data buffering
+ Once buffering has started (with an mdat atom), continue buffering
+ until moov atom is reached, which handles cases with multiple
+ mdat atoms. Also keep adapter/offset better in sync with upstream
+ and fix some debug statements. Fixes #587426.
+
+2009-07-06 10:40:31 +0200 Philip Jägenstedt <philipj@opera.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Replace deprecated GST_DISABLE_DEBUG with correct macro. Fixes #587826
+
+2009-07-01 13:07:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: error out instead of dividing by 0
+ Error out if timescale is 0.
+
+2009-07-01 09:32:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ Revert "qtdemux: Make sure we don't blacklist streams by wrongly comparing their"
+ This reverts commit 5503a59a5779b67451d8a271000181790ee76bc7.
+ Reverting this since it causes regressions with a lot of sample files
+ I have, all of which worked fine with the last -good release (#586891).
+
+2009-06-30 15:54:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: comment out unused structure
+
+2009-06-30 13:12:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: more size checks, and use g_try_new0() instead of g_new0()
+ Whenever we alloc something based on a user-supplied size, we should
+ really use g_try_new(), otherwise we can easily be made to abort by
+ passing a ridiculously large number to us for allocing. Fixes
+ problems with some fuzzed files.
+
+2009-06-29 18:58:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: guard against bogus atom sizes and short reads
+ Check the possibly 64-bit atom size more carefully before casting it
+ to an int and passing it to gst_pad_pull_range(), otherwise we might
+ end up pulling 0 bytes, getting an empty buffer as requested and
+ dereferencing not available data whilst thinking we actually asked
+ for and got 0x1000000000000 bytes. Similar fix for push mode operation
+ where neededbytes ends up being 0 bytes, which makes us assert. Fixes
+ crash with broken or fuzzed file (NB #122378).
+
+2009-06-29 16:52:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: use 0x prefix when logging numbers in hex
+
+2009-07-01 08:40:40 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: Don't send empty string tags
+
+2009-06-30 21:35:37 +0400 LRN <lrn1986 at gmail.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ Don't use sendmsg()-dependent code on Windows
+ Fixes #585842
+
+2009-06-30 15:59:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/law/alaw-decode.c:
+ * gst/law/alaw-encode.c:
+ * gst/law/alaw.c:
+ * gst/law/mulaw-decode.c:
+ * gst/law/mulaw-encode.c:
+ * gst/law/mulaw.c:
+ law: fix caps and negotiation
+ Fix the caps to include the depth (instead of width twice) in the caps of
+ audio/x-raw-int.
+ Fix negotiation to not only copy the rate/channels of the first structure.
+
+2009-06-30 14:48:09 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: include "1.0=100%" in volume and change upper limit
+ Upper volume limmit was 1000. That appear unneceasrily high. It would also cause
+ sever distortion if accidentialy used. Now its 10 (~ +15db) which is also in
+ sync with volume and playbin2.
+
+2009-06-29 15:39:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesrc.c:
+ pulse: some more trivial cleanups
+
+2009-06-29 15:38:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsemixer.c:
+ pulse: trivial cleanups
+
+2009-06-29 15:20:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: clear ringbuffer when asked to
+ Since we map the ringbuffer to the pulseaudio internal ringbuffer, flush the
+ pulseaudio buffer when we are asked to clear the ringbuffer.
+ This avoids some leftover audio after a seek.
+
+2009-06-26 15:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * autogen.sh:
+ autogen.sh: Actually do the 'echo -n' -> printf change.
+
+2009-06-26 14:40:14 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * autogen.sh:
+ autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
+ Check for more automake command variants. Use printf instead of 'echo -n'
+ for portability
+
+2009-06-26 13:42:09 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
+ Automatic update of common submodule
+ From f810030 to 5845b63
+
+2009-06-26 13:19:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: don't process track_num/track_count tags with a 0 value
+ Number/count values of 0 mean they're not set. Don't put those in the
+ taglist.
+
+2009-06-25 18:51:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/waveform/gstwaveformsink.c:
+ waveformsink: use 'guint8' instead of 'byte' to fix compilation with MSVC8
+ We need a cast here for pointer arithmetic to work correctly, but some
+ MSVC versions don't seem to like 'byte', so use guint8 here. Hopefully
+ fixes #585361.
+
+2009-06-25 19:39:37 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2src: set structs to zero before using them in ioctls
+ This fixes valgrind warnings.
+
+2009-06-25 13:23:40 +0200 Julien Moutte <julien@fluendo.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Make sure we don't blacklist streams by wrongly comparing their duration with entire clip duration.
+
+2009-06-25 13:18:14 +0200 Krzysztof Błaszkowski <kb at sysmikro.com.pl>
+
+ * gst/rtsp/gstrtpdec.c:
+ rtpdec: fix some buffer leaks
+
+2009-06-25 08:11:09 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/flv/gstflvparse.c:
+ flvparse: Add missing break in switch/case.
+
+2009-06-25 08:10:38 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Remove unused variable, hint branch likeliness, add comments.
+
+2009-06-25 08:09:57 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Removed unused variable
+
+2009-06-25 07:41:07 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Remove dead assignments and unused variables.
+ Also add branch likeliness macros.
+
+2009-06-25 07:40:26 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Fix uninitialized variables. Fixes build on macosx
+
+2009-06-24 17:43:25 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: free memory in finalize
+ finalize is called only once. no need to clear pointers there. dispose is for
+ unreffing.
+
+2009-06-24 15:14:14 +0100 Jan Schmidt <jan.schmidt@sun.com>
+
+ * common:
+ Automatic update of common submodule
+ From 6ab11d1 to f810030
+
+2009-06-08 14:46:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: short-circuit gst_avi_demux_src_convert() when parsing the index
+ Don't call gst_avi_demux_src_convert() for each single index entry. Not
+ only do we already have the pointer to the stream context, we also know
+ the formats we want to convert from and to already, so we may just as
+ well use optimised conversion routines that bypass some of the checks
+ and lookups made in gst_avi_demux_src_convert().
+
+2009-06-17 16:39:36 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Another round of G_*LIKELY micro-optimisations.
+
+2009-06-17 16:20:25 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Take last sample duration for dummy segment calculation.
+ This fixes the cases where files without EDL wouldn't output their
+ last buffer.
+
+2009-06-24 12:36:31 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Sprinkle branch likeliness macros over the code.
+
+2009-06-23 16:54:32 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/raw1394/gstdv1394src.c:
+ * ext/raw1394/gsthdv1394src.c:
+ raw1394: sprinkle branch likeliness macros accross the code.
+
+2009-06-14 10:36:17 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Add GST_MEMDUMP statements for unknown atoms.
+ This is to help developers track down and implement unhandled atoms faster.
+
+2009-06-23 17:51:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Remove the interlaced field from the output caps if deinterlacing is enabled
+
+2009-06-23 17:48:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/tvtime/greedyh.c:
+ deinterlace: Copy the correct line from correct place in the history
+
+2009-06-23 16:35:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: use same protocols after redirect
+ After a redirect we want to use the same protocols that we were using for the
+ current url.
+
+2009-06-23 15:35:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: don't leak cover art
+
+2009-06-23 14:10:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/udp/gstudpnetutils.c:
+ udp: fix compiler warning about EAI_ADDRFAMILY getting redefined in some cases
+ Include the header from where we include all the system headers with the
+ socket stuff before we try to define EAI_ADDRFAMILY ourselves, otherwise
+ we define it ourselves and then get a compiler warning if a system header
+ defines it as well without guarding against it being defined already.
+
+2009-06-23 14:39:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/matroska/matroska-ids.h:
+ matroska: and the new headers too
+
+2009-06-23 14:32:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroske: fix compiler error
+ change gpointer to guint8 * for codec_state and codec_priv as some
+ functions operate on those types and it avoids breaking strict-aliasing
+ rules.
+
+2009-06-23 12:42:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: avoid leaking buffers
+ Don't leak buffers when resyncing to a keyframe.
+ Avoid leaking buffers when exiting the loop on error conditions.
+ Add some more debug info.
+ Fixes #585911
+
+2009-06-22 15:56:58 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: open/close the device in READY
+ This allows to query the device in READY. Before one need to switch it to PAUSED
+ and that also starts streaming.
+
+2009-06-20 15:41:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_dump.c:
+ qtdemux: use GST_MEMDUMP
+
+2009-06-19 00:16:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/apetag/Makefile.am:
+ * gst/apetag/gstapedemux.c:
+ apedemux: add container-format tag
+ Use pbutils here because the string is translated.
+
+2009-06-19 00:15:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/id3demux/Makefile.am:
+ * gst/id3demux/gstid3demux.c:
+ id3demux: add container-format tag
+ Using pbutils here because the string is translated.
+
+2009-06-18 23:51:52 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: post container-format tag
+ Also merge the two almost identical _add_*_pad() functions into one.
+
+2009-06-18 23:43:49 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: don't screw up first audio buffer
+ Query the audio format, esp. dvdemux->num_channels, before we use that
+ variable to allocate the initial buffer. That way we don't accidentally
+ push a zero-sized buffer as first audio buffer.
+
+2009-06-18 23:38:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: post container-format tag
+
+2009-06-18 23:37:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: post container-format tags
+
+2009-06-18 23:36:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: post container-format tag
+
+2009-06-18 23:35:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: post container-format tags
+
+2009-06-21 17:13:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audioamplify.c:
+ audioamplify: Fix integer overflows on 32 bit architectures
+
+2009-06-21 09:50:54 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
+
+ * gst/audiofx/audioamplify.c:
+ audioamplify: Don't declare a loop index static
+ The previous patch to add support for additional sample formats possibly
+ introduced a reentrancy bug: a variable used for a loop index was declared
+ static. This patch fixes that, and also adds a "/* *INDENT-ON* */" annotation
+ following the macro block. (I don't know what the annotation is for, but the
+ adder, where I copied this from, has it).
+
+2009-06-19 22:37:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audioamplify.c:
+ audioamplify: Fix off-by-one in wrap-positive mode
+
+2009-06-19 22:20:45 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
+
+ * gst/audiofx/audioamplify.c:
+ * gst/audiofx/audioamplify.h:
+ audioamplify: Add noclip method and support for more formats
+ Fixes bug #585828 and #585831.
+
+2009-06-19 21:46:41 +0200 Koop Mast <kwm@freebsd.org>
+
+ * gst/udp/gstudpnetutils.h:
+ udp: Fix build on FreeBSD
+ Fixes bug #586397.
+
+2009-06-19 18:12:27 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: add unit tests for buffer-list payloaders
+ See #585559
+
+2009-06-19 18:00:35 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmp4vpay.h:
+ rtpmp4vpay: add support for buffer-list
+ See #585559
+
+2009-06-19 17:57:12 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpjpegpay.h:
+ rtpjpegpay: add support for buffer-lists
+ See #585559
+
+2009-06-19 17:53:32 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ rtph264pay: add support for buffer-lists
+ See #585559
+
+2009-06-18 11:54:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/udp/gstudpnetutils.c:
+ udputils: don't free invalid memory
+ As spotted by benjiG in IRC.
+ don't free invalid memory when getaddrinfo failed.
+
+2009-06-17 17:48:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulseink: don't leak device_description
+ don't leak the device_description.
+ some cleanups.
+
+2009-06-19 14:44:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ po: update .po files for sunaudiomixer string changes
+
+2009-06-18 16:58:26 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: streaming; adjust sizes to cater for padding in chunks
+
+2009-06-17 11:54:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: streaming mode; handle data chunks grouped in rec lists.
+ Fixes #567983.
+
+2009-06-10 12:36:50 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: map some tags to COMPOSER rather than ARTIST
+
+2009-06-10 12:34:43 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix some 3GP tag extraction (keywords, genre, location)
+
+2009-06-09 15:36:50 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_fourcc.h:
+ qtdemux: extract pixel-aspect-ratio information
+
+2009-06-17 07:14:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Fix leaking of the Matroska TITLE element
+
+2009-06-16 20:38:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-annodex.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-efence.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-esdsink.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gamma.xml:
+ * docs/plugins/inspect/plugin-gconfelements.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-halelements.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-monoscope.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-quicktime.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobalance.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videoflip.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstaging.h:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gstdice.h:
+ * gst/effectv/gstedge.c:
+ * gst/effectv/gstedge.h:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstquark.h:
+ * gst/effectv/gstrev.c:
+ * gst/effectv/gstrev.h:
+ * gst/effectv/gstshagadelic.c:
+ * gst/effectv/gstshagadelic.h:
+ * gst/effectv/gstvertigo.c:
+ * gst/effectv/gstvertigo.h:
+ * gst/effectv/gstwarp.c:
+ * gst/effectv/gstwarp.h:
+ effectv: Add basic documentation for the effectv elements
+
+2009-06-16 20:16:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gsteffectv.h:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstshagadelic.c:
+ effectv: Define the fast PRNG function at a central place
+
+2009-06-16 20:13:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/Makefile.am:
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstaging.h:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gstdice.h:
+ * gst/effectv/gstedge.c:
+ * gst/effectv/gstedge.h:
+ * gst/effectv/gsteffectv.c:
+ * gst/effectv/gsteffectv.h:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstquark.h:
+ * gst/effectv/gstrev.c:
+ * gst/effectv/gstrev.h:
+ * gst/effectv/gstshagadelic.c:
+ * gst/effectv/gstshagadelic.h:
+ * gst/effectv/gstvertigo.c:
+ * gst/effectv/gstvertigo.h:
+ * gst/effectv/gstwarp.c:
+ * gst/effectv/gstwarp.h:
+ effectv: Move type definitions into separate headers
+ This is needed for the docs later.
+
+2009-06-16 19:41:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gstedge.c:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstrev.c:
+ * gst/effectv/gstshagadelic.c:
+ * gst/effectv/gstvertigo.c:
+ * gst/effectv/gstwarp.c:
+ effectv: Remove get_unit_size implementations
+ The default on from GstVideoFilter handles this already.
+
+2009-06-16 14:54:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to git
+ Need git core for basesink bufferlist additions; -base requirement
+ bumped gratuitously.
+
+2009-06-16 15:25:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/udpsink.c:
+ tests: add some debug, send newsegment
+
+2009-06-16 15:06:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: add debug line for the socket
+
+2009-06-16 15:06:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/pipelines/flacdec.c:
+ tests: turn g_print into debug
+
+2009-06-16 15:04:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/udpsink.c:
+ multiudpsink: add support for buffer lists
+ Add support for BufferList and add a unit test.
+ Fixes #585842
+
+2009-06-16 00:02:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: reset session state when stopping
+ Increases the chances that the element is actually reusable.
+
+2009-06-15 23:49:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: log response and request headers and fix some broken indenting
+
+2009-06-15 22:40:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpmp4gdepay.c:
+ mp4gdepay: guess constantDuration better
+ Do a better job at guessing the constantDuration parameter when it is not
+ present in the caps.
+ Fixes #585205
+
+2009-06-15 21:09:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstwarp.c:
+ warptv: Clean up warptv element and fix some minor bugs and leaks
+
+2009-06-15 20:53:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstvertigo.c:
+ vertigotv: Clean up vertigotv element and fix some minor bugs and leaks
+
+2009-06-15 20:38:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstdice.c:
+ dicetv: Use guint8 instead of char (which can be signed or unsigned)
+
+2009-06-15 20:36:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstshagadelic.c:
+ shagadelictv: Use guint8/gint8 instead of char (which can be signed or unsigned)
+
+2009-06-15 20:31:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstshagadelic.c:
+ shagadelictv: Clean up element and free all memory in finalize
+
+2009-06-15 20:21:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstrev.c:
+ revtv: Clean up revtv element
+
+2009-06-15 20:07:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstquark.c:
+ quarktv: Simplify some code
+
+2009-06-15 20:07:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstquark.c:
+ quarktv: Use the input data if a NULL buffer is chosen instead of the value 0
+
+2009-06-15 20:00:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstquark.c:
+ quarktv: Fix setting the planes property of quarktv
+ Setting it to a value<16 would cause crashes before because
+ current_plane was set to the old number of planes-1. Also
+ fix calculations for non-2^n planes values.
+
+2009-06-15 17:50:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstquark.c:
+ quarktv: Clean up the quarktv element
+
+2009-06-15 17:39:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gsteffectv.c:
+ effectv: Make elements list constant
+
+2009-06-15 17:37:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstedge.c:
+ edgetv: Clean up edgetv element and fix memory leak
+
+2009-06-15 17:21:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstdice.c:
+ dicetv: Clean up dicetv element and fix some smaller issues
+ This fixes a memory leak (the dice map) and a crash when
+ setting the square-bits property before caps are set.
+
+2009-06-15 17:20:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/Makefile.am:
+ * gst/effectv/gstaging.c:
+ agingtv: Actually use GstController for syncing the properties to timestamps
+
+2009-06-15 17:03:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstaging.c:
+ agingtv: Export some more agingtv properties via GObject properties
+
+2009-06-15 15:06:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstaging.c:
+ agingtv: General cleanup and updating of copyright
+ Also make the scratch-lines property exported via a GObject
+ property and initialize/reset the internal state correctly.
+
+2009-06-15 15:05:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/effectv/gstaging.c:
+ agingtv: Store and update state inside the instance struct
+ This makes the coloraging effect and pits effect visible.
+
+2009-06-15 15:51:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: ref custom ring buffer class and type in class_init
+ Hack around thread-safety issues in GObject and our racy _get_type()
+ functions (we could easily fix the _get_type() functions, but we still
+ need to hack around the GObject class races until we require a newer
+ GLib version, I think).
+
+2009-06-14 19:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/dv/demo-play.c:
+ * tests/old/examples/Makefile.am:
+ * tests/old/examples/level/Makefile.am:
+ * tests/old/examples/level/README:
+ * tests/old/examples/level/demo.c:
+ * tests/old/examples/level/plot.c:
+ * tests/old/examples/switch/.gitignore:
+ * tests/old/examples/switch/Makefile.am:
+ * tests/old/examples/switch/switcher.c:
+ Remove a few old example apps from the 0.8 days
+ Some have been replaced by newer ones, others are demoing elements that
+ don't exist any longer (not in -good anyway), and others have not been
+ touched in many years and it seem pointless to keep them around.
+ Removing these files makes sure we don't have any code in our repository
+ that uses Gtk+ symbols which are to be removed for GNOME3, and as such
+ will make some script that greps for this kind of stuff give us a clean
+ bill of code health. Fixes #585757.
+
+2009-06-13 21:02:45 -0400 Olivier Crête <tester@tester.ca>
+
+ * common:
+ * gst/rtp/gstrtpsirenpay.c:
+ rtpsirenpay: Remove deprecated symbol
+ Patch by: Luis Menina
+
+2009-06-13 10:43:55 +0200 Marvin Schmidt <marvin_schmidt@gmx.net>
+
+ * tests/check/Makefile.am:
+ tests: Don't run the flacdec test if the plugin isn't built. Fixes #585630
+
+2009-06-12 16:06:28 +0200 Patrick Radizi <patrick.radizi at axis.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Add RTP blocksize functionality
+ Add property to make the client suggest a blocksize to the server.
+ Fixes #585549
+
+2009-06-11 22:30:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/README:
+ rtp: update README, fix some typos, mention gstrtpbin
+
+2009-06-11 19:10:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: handle border cases in resampler
+
+2009-06-11 13:32:22 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
+ * docs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ * docs/upload.mak:
+ docs: Bump common. Use upload-doc.mak instead of upload.mak
+ Remove the local copy of upload.mak in favour of using the shared
+ upload-doc.make in common/
+
+2009-06-11 11:39:25 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/goom/goom_config_param.h:
+ * gst/videomixer/videomixer.c:
+ docs: Quieten a couple more docs warnings
+
+2009-06-11 11:27:26 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/matroska/lzo.c:
+ docs: Remove gtk-doc comment marker
+ These comment blocks aren't gtk-doc comments and cause annoying noise in
+ the docs build.
+
+2009-06-11 10:05:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Implement upstream negotation
+
+2009-06-10 21:47:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Improve debugging and clean up some code
+
+2009-06-10 14:55:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Clip buffers to the current segment if possible
+
+2009-06-10 14:45:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Clean up includes and clean up order of instance struct fields
+
+2009-06-10 16:09:56 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/gstrtph263pay.h:
+ rtph263pay: Default to doing A, B and C modes, not only A
+
+2009-06-10 09:56:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Fix QoS calculations
+ The diff is a signed integer, not an unsigned one of course.
+ In modes other than GST_DEINTERLACE_ALL every frame has twice the
+ duration of the field duration.
+
+2009-06-09 14:13:31 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/gstrtpsirenpay.c:
+ rtpsirenpay: Put the bitrate in the RTP caps
+ The MS code seems to require the bitrate to interoperate and
+ draft-ietf-avt-rtp-g7221-00 also has it.
+
+2009-06-09 19:55:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Implement basic QoS
+ This change is based on Tim's QoS implementation
+ for jpegdec.
+
+2009-06-09 19:29:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Directly proxy events/queries to the peer pads
+ This removes some overhead introduced by the default handlers
+ that need to iterate over the other pads.
+
+2009-06-09 10:38:52 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: debug_memdump() unknown tags. Refactor junk parsing code.
+ This makes life slightly easier when debugging avi files.
+
+2009-06-08 08:21:43 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/Makefile.am:
+ rtp: Don't forget to dist the headers for the CELT (de)payloaders.
+
+2009-06-07 20:54:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ Revert "Revert "qtdemux: fill timestamp table completely""
+ This reverts commit 9f022c8a8503c2ce0fa617fdb50e41706dd412f5.
+ Sorry, I was thinking about the wrong module.
+
+2009-06-07 20:49:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ Revert "qtdemux: fill timestamp table completely"
+ This reverts commit 790b050fc5302cae89cddcd23b258093967d05a9.
+ I forgot we were frozen.
+
+2009-06-07 20:46:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fill timestamp table completely
+ When there are less timestamps that there are samples, fill up the sample table
+ with the last know timestamp. This situation can happen when the last sample
+ does not decode and doesn't need a timestamp. We however calculate the total
+ track length using the last sample timestamp so we need to have something
+ sensible in there.
+ Fixes #585056
+
+2009-06-07 13:37:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: handle LIST INFO of 0 size
+ Handle LIST INFO chunks of 0 size instead of causing errors.
+ Fixes #584981
+
+2009-06-07 13:24:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/wavparse/gstwavparse.c:
+ Revert "wavparse: Remove dead assignments, move variable to where it's needed."
+ Reverts commit 44256a78f8dd79a91f3bb2ab7c3aa623c097bb8a and use the result in
+ error reporting so that we can see what's going on.
+
+2009-06-05 18:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltdepay.h:
+ celtdepay: add CELT depayloader
+
+2009-06-05 15:30:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpceltpay.h:
+ rtpceltpay: add CELT RTP payloader
+
+2009-06-05 16:54:48 +0100 Jan Schmidt <jan.schmidt@sun.com>
+
+ * sys/sunaudio/gstsunaudiomixerctrl.c:
+ * sys/sunaudio/gstsunaudiomixeroptions.c:
+ * sys/sunaudio/gstsunaudiomixertrack.c:
+ sunaudio: Fix switch setting on some devices. Add debug. Fix a FIXME.
+ Fix the setting of toggle switches on some broken audio drivers which
+ report that no audio ports are settable by ignoring the mod_port field
+ there.
+ Add some debug statements.
+ Fix a FIXME now that Good relies on a new enough gst-plugins-base.
+
+2009-06-04 12:27:19 +0100 Jan Schmidt <jan.schmidt@sun.com>
+
+ * sys/sunaudio/Makefile.am:
+ * sys/sunaudio/gstsunaudiomixerctrl.c:
+ * sys/sunaudio/gstsunaudiomixerctrl.h:
+ * sys/sunaudio/gstsunaudiomixeroptions.c:
+ * sys/sunaudio/gstsunaudiomixeroptions.h:
+ * sys/sunaudio/gstsunaudiomixertrack.c:
+ * sys/sunaudio/gstsunaudiomixertrack.h:
+ sunaudio: Support new flags for options and actions
+ Use new audio mixer flags added in Base 0.10.23 to expose flags and options
+ on the SunAudio devices.
+ Fixes: #583593
+ Patch By: Brian Cameron <brian.cameron@sun.com>
+ Patch By: Garrett D'Amore <garrett.damore@sun.com>
+
+2009-05-15 11:50:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: First try to handle DVD still frames correctly
+ This helps a bit with bug #582740 but still doesn't make it work.
+
+2009-06-04 17:37:03 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: only notify if all checks passed
+ Replace goto done: with return, as those are checks when we don't want to flag a
+ pending notify.
+
+2009-06-04 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: set the right state on rtpbin
+ We need to set the state of gstrtpbin to the same state as our source elements.
+ This fixes fallback to TCP again.
+
+2009-06-03 18:23:53 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: check pointer before accessing
+ Move existing check a few lines up, so that we check before accessing fields.
+
+2009-06-03 18:21:12 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: rename gst_pulse_sink_get_time to gst_pulsesink_get_time
+ Rename internal method for consistency.
+
+2009-06-03 18:19:22 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: use values from pa_stream_get_buffer_attr()
+ We were putting the requested values back into ringbuffer spec, instead of
+ using the queried values.
+
+2009-06-02 19:32:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpvrawpay.c:
+ vrawpay: trim output buffers
+ Remove the leftover unused bytes in the output buffer.
+ Fixes #584613
+
+2009-06-02 19:30:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ vrawdepay: fix parsing of sampling field
+ commit a12d9a80f225be97b3674b1a0506ac66544dbf49 broke the parsing of the
+ sampling.
+
+2009-05-27 17:06:34 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: Avoid possible overflow in calculations
+ A malformed (or simply huge) PNG file can lead to integer overflow in
+ calculating the size of the output buffer, leading to crashes or buffer
+ overflows later. Fixes SA35205 security advisory.
+
+2009-06-02 00:48:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: some more logging - dump header packets
+ Also, the final fixing up of the headers is expected and not something
+ we should warn about.
+
+2009-06-02 00:37:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: never ever pass values >36bits to _set_total_samples_estimate()
+ Let's be paranoid and make sure we never pass a number that takes up
+ more than 36 bits to _set_total_samples_estimate(), since libFLAC
+ expects all the other bits to be zero, and if this is not the case
+ neighbouring fields in the global stream info header may get messed
+ up inadvertently, so that flac -d refuses to decode the stream.
+ See #584455.
+
+2009-06-01 22:33:02 +0200 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+
+ * ext/flac/gstflacenc.c:
+ Address bad FLAC sample length encoding of #5844455
+ Commit df707c666433a78d3878af6f055698d5756226c4
+ introduced an obvious bug in the sample length calculation,
+ using the wrong macro for conversion.
+
+2009-06-01 11:58:21 -0700 Brian Cameron <brian.cameron@sun.com>
+
+ * gst/deinterlace/tvtime/mmx.h:
+ deinterlace: Fix spurious colons in asm code
+ Fixes #584174.
+ Signed-off-by: David Schleef <ds@schleef.org>
+
+2009-06-01 00:40:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: skip JUNK chunks in data section in streaming mode
+ Skip JUNK tags in streaming mode as well instead of EOSing
+ prematurely. Fixes #564100.
+
+2009-05-28 14:01:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/blend_i420.c:
+ * gst/videomixer/videomixer.c:
+ videomixer: Don't use // comments
+
+2009-05-28 13:56:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_bgra.c:
+ videomixer: Fix background blitting when a color mode is selected with BGRA
+
+2009-05-28 13:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/blend_i420.c:
+ * gst/videomixer/videomixer.c:
+ * gst/videomixer/videomixer.h:
+ videomixer: Some cleanup and fix the calculation of the frame size in bytes
+
+2009-05-28 13:35:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_i420.c:
+ videomixer: Fix I420 blending to actually do something
+ For this we a) implement the checkers filling and b)
+ actually blend the src/dest by using the src alpha value
+ from the pad.
+
+2009-05-28 13:14:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_bgra.c:
+ videomixer: Fix ARGB blending to actually work
+
+2009-05-28 13:04:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/blend_bgra.c:
+ videomixer: Blend BGRA ourselves instead of using Cairo
+
+2009-05-28 12:55:16 +0200 Alex Ugarte <alexugarte@gmail.com>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/blend_i420.c:
+ * gst/videomixer/videomixer.c:
+ * gst/videomixer/videomixer.h:
+ videomixer: Add support for blending BGRA and AYUV
+ Fixes bug #577017.
+
+2009-05-28 12:39:46 +0200 Ghislain 'Aus' Lacroix <aus@songbirdnest.com>
+
+ * gst/equalizer/gstiirequalizer.c:
+ equalizer: Use floating point arithmetic internally for the int16 mode
+ By using int32 arithmetic we will introduce distortions as the
+ IIR filter is very sensitive to rounding errors. Fixes bug #580214.
+
+2009-05-28 10:55:16 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
+
+ * gst-plugins-good.spec.in:
+ Update spec file with latest plugins
+
+2009-05-26 17:19:08 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
+ Automatic update of common submodule
+ From 888e0a2 to c572721
+
+2009-05-26 16:20:35 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2: cleanup and commenting
+ Remove newlines inserted by gst-indent once. Remove unused var from instance
+ struct. Add comments. Add another #define for default property value.
+
+2009-05-06 12:43:35 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/check/Makefile.am:
+ makefile: idea about makeing more sources/sinks testable again
+
+2009-05-25 16:33:35 +0200 John Keeping <john.keeping at lineone.net>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: match g_malloc() with g_free()
+ Matching g_malloc() with a g_free() is important when a custom allocator is
+ installed.
+ Fixes #583803
+
+2009-05-12 18:39:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmp4vpay.h:
+ rtpmp4vpay: don't look for headers in some cases
+ In some streams (starting with 00000100) don't look for the headers but push
+ data as it is.
+ Fixes #582153
+
+2009-05-13 11:50:22 +0200 Patrick Radizi <patrick.radizi at axis.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix memory leak of messages
+ Free messages correctly.
+ Fixes #577318
+
+2009-05-24 19:32:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: make fakesrc silent
+ Make the fakesrc that is responsible for sending dummy packets silent.
+
+2009-05-24 16:33:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: don't send teardown before setup
+ Don't send a TEARDOWN request when we did not manage to successfully setup a
+ stream.
+
+2009-05-14 14:46:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: Populate a GstIndex that is set on matroskademux
+
+2009-05-14 10:35:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Get the max duration from upstream if there's no duration tag
+
+2009-05-14 10:29:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Write an index table to the end of the file
+
+2009-05-22 01:12:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * autogen.sh:
+ * configure.ac:
+ autotools: move the -Wno-portability from autogen.sh to configure.ac
+ If we're lucky it'll get used on automatic rebuilds as well that way.
+
+2009-05-22 01:10:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ * configure.ac:
+ * m4/gst-fionread.m4:
+ m4: fix 'suspicious cache id' warnings
+ and update common to pull in a similar fix. Also check in configure
+ whether the compiler supports do while macros (GLib wants this
+ defined and it is needed to avoid warnings with some c++ compilers
+ apparently).
+
+2009-05-22 01:39:33 +0300 Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
+
+ * configure.ac:
+ souphttpsrc: Bump-up libsoup-2.24 dep to >= 2.26
+ The helper function soup_message_headers_get_content_type that we now use
+ was added in 2.26.
+
+2009-05-20 17:57:59 +0300 Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Set caps for audio/L16 content-type
+ When "Content-Type" header is "audio/L16", we need to set the caps on the
+ outgoing buffers so that downstream elements can have means to detect the
+ stream type and handle it appropriately. Tested with HTTP stream provided
+ by pulse-audio's http module (git master).
+
+2009-05-20 15:06:25 +0300 Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: Rename icy_caps to src_caps
+
+2009-05-21 23:39:13 +0200 Philippe Normand <philippe at fluendo.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: bump max size to 65535x65535
+ Remove artificial jpeg image limits.
+ Fixes #583048.
+
+2009-05-21 21:36:02 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * win32/common/config.h:
+ win32: Update the win32 config.h
+
+2009-05-19 15:12:09 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: Recognise PGS subpicture streams - the bluray format.
+ Recognise and apply appropriate caps to PGS (Presentation Graphic Stream)
+ subpicture streams.
+
+2009-05-15 10:42:19 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Convert an erroneous assertion
+ Occasionally, we get a change callback for an old stream, triggering
+ the assertion unnecessarily. Just ignore such callbacks.
+
+2009-05-20 16:14:40 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulse: Print a warning on under/overflows
+
+2009-05-20 18:45:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_fourcc.h:
+ qtdemux: parse in24 boxes to get endianness
+ in24 samples are normally big-endian but an enda box can change this to
+ little-endian. Recurse into the in24 box and find the enda box so that we get
+ the endianness right.
+ Fixes #582515
+
+2009-05-20 14:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: add proper padtemplate
+
+2009-05-20 14:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: add more mime types
+ Add mime-type for Panasonic g726 and add more required caps properties for other
+ G726 mime-types.
+ Make mime-types case insensitive.
+ See #582169
+
+2009-05-20 13:47:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/multipart/multipartdemux.c:
+ * gst/multipart/multipartdemux.h:
+ multipartdemux: add flow aggregation
+
+2009-05-20 13:29:02 +0200 Arnout Vandecappelle <arnout@mind.be>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: allow content to be empty.
+ gst_adapter_take_buffer doesn't allow buffer to be empty.
+ Simply skip any part where the content is empty. Don't
+ create a pad for it either.
+ See #582169
+
+2009-05-18 22:19:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpchannels.h:
+ rtp: fix channel positions for mono
+
+2009-05-21 21:02:11 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * configure.ac:
+ Back to hacking -> 0.10.15.1
+
=== release 0.10.15 ===
-2009-05-20 Jan Schmidt <jan.schmidt@sun.com>
+2009-05-20 22:34:18 +0100 Jan Schmidt <thaytan@noraisin.net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 0.10.15, "I've been up all night"
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-annodex.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-efence.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-esdsink.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gamma.xml:
+ * docs/plugins/inspect/plugin-gconfelements.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-halelements.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-monoscope.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-quicktime.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobalance.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videoflip.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 0.10.15
+
+2009-05-20 22:03:21 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
2009-05-16 02:59:14 +0100 Jan Schmidt <thaytan@noraisin.net>
* tests/check/Makefile.am:
* tests/check/audiotestsrc.flac:
* tests/check/pipelines/flacdec.c:
- add a test to check that we get all decoded bytes
- from a 10-buffer audiotestsrc flac, in the case of:
- - a full decode
- - a decode of a seek for the full file
- - a decode of a seek for a small part, smaller than the first buffer
+ add a test to check that we get all decoded bytes from a 10-buffer audiotestsrc flac, in the case of: - a full decode - a decode of a seek for the full file - a decode of a seek for a small part, smaller than the first buffer
The test fails because flacdec drops the first outgoing buffer on a seek
2009-03-03 10:06:52 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
* ext/pulse/pulsesink.c:
* ext/pulse/pulsesink.h:
- pulsesink: Issue property change notification in streaming thread,
- rather than PA thread.
+ pulsesink: Issue property change notification in streaming thread, rather than PA thread.
pa_threaded_mainloop_lock() (a.o.) and by extension get_property should
not be done from a PA thread, but the latter may occur as a result of a
property change notification. Fixes #571204 (though current situation
* ext/pulse/pulsesrc.h:
* ext/pulse/pulseutil.c:
* ext/pulse/pulseutil.h:
- Rewrite the pulse plugin, conditionally enabling new behaviour with
- newer pulseaudio.
+ Rewrite the pulse plugin, conditionally enabling new behaviour with newer pulseaudio.
Fixes: #567794
* Hook pulsesink's volume property up with the stream volume -- not the
sink volume in PA.
projects / platform / upstream / gstreamer.git / commitdiff