+=== release 1.1.4 ===
+
+2013-08-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ releasing 1.1.4
+
+2013-08-28 12:32:10 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * po/pt_BR.po:
+ po: update translations
+
+2013-08-27 15:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: remove framerate restriction
+ Remove the framerate restriction on the caps.
+
+2013-08-27 09:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ session: only update next check time when reconsidering
+ Don't update the next RTCP check time in all cases but only when we
+ reconsidered. This avoids delaying sending a full RTCP packet when we
+ are doing early feedback.
+
+2013-08-27 09:37:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ session: add more debug
+
+2013-08-27 09:34:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ jitterbuffer: fix types of the retransmission event
+
+2013-08-27 09:33:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: only timeout EXPECTED timers on gap
+ Only timeout the EXPECTED timers when we detect a large seqnum gap.
+
+2013-08-26 13:47:53 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ configure.ac: Don't set BZ2_LIBS if bz2 is not found
+
+2013-08-26 11:50:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtsession: fix locking
+ We need to take the session lock when getting and manipulating the
+ source.
+
+2013-08-26 11:50:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: add some more debug
+
+2013-08-20 22:12:03 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: don't send flush_stop twice.
+ If we get flush start and a seek we need to only send flush_stop once.
+ More info at #706441
+
+2013-08-23 15:56:43 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/multipart/multipartdemux.c:
+ * gst/multipart/multipartdemux.h:
+ multipartdemux: propagate discont
+
+2013-08-23 15:49:47 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: remove dynamic sourcpads when going from PAUSED to READY
+
+2013-08-23 15:29:28 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/multipart/multipartdemux.c:
+ * gst/multipart/multipartdemux.h:
+ multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last
+ https://bugzilla.gnome.org/show_bug.cgi?id=637754
+
+2013-08-23 15:47:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxqueue.h:
+ rtxqueue: add property to configure queue size
+
+2013-08-23 12:07:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/rtp/client-H264-rtx.sh:
+ * tests/examples/rtp/server-VTS-H264-rtx.sh:
+ tests: add retransmission example
+
+2013-08-23 11:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: proxy jitterbuffer do-retransmission property
+
+2013-08-23 11:17:45 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/avi/gstavimux.c:
+ avimux: unmap the correct buffer
+ The audio buffer was mapped so unmap it and not the video buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=706642
+
+2013-08-18 23:32:22 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesink.h:
+ pulsesink: Add property to find out the device currently in use
+ https://bugzilla.gnome.org/show_bug.cgi?id=590768
+
+2013-08-18 23:31:15 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: De-duplicate code to get the current sink input info
+ https://bugzilla.gnome.org/show_bug.cgi?id=590768
+
+2013-08-18 22:27:37 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Implement changing the device while playing
+ https://bugzilla.gnome.org/show_bug.cgi?id=590768
+
+2013-08-18 23:32:22 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/pulsesrc.c:
+ * ext/pulse/pulsesrc.h:
+ pulsesrc: Add property to find out the device currently in use
+ https://bugzilla.gnome.org/show_bug.cgi?id=590768
+
+2013-08-18 23:31:15 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: De-duplicate code to get the current source output info
+ https://bugzilla.gnome.org/show_bug.cgi?id=590768
+
+2013-08-18 22:27:37 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: Implement changing the device while playing
+ https://bugzilla.gnome.org/show_bug.cgi?id=590768
+
+2013-08-22 14:55:14 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ configure: Fix bz2 configure check for Windows
+ Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2.
+ https://bugzilla.gnome.org/show_bug.cgi?id=465924
+
+2013-02-22 20:57:00 +0900 Akihiro Tsukada <atsukada@users.sourceforge.net>
+
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesink.h:
+ * ext/pulse/pulseutil.c:
+ * ext/pulse/pulseutil.h:
+ pulsesink: Add support for AAC pass-through
+ https://bugzilla.gnome.org/show_bug.cgi?id=694445
+
+2013-06-24 17:29:37 +0200 Kishore Arepalli <kishore.arepalli@gmail.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ gdkpixbufoverlay: crashes if any property changes during playback when location property is not set
+ https://bugzilla.gnome.org/show_bug.cgi?id=702988
+
+2013-08-21 14:54:26 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesink.h:
+ * ext/pulse/pulsesrc.c:
+ * ext/pulse/pulseutil.h:
+ pulse: Share static caps definition between src and sink
+ The src was also missing 24-bit sample formats
+
+2013-08-21 16:53:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxqueue.h:
+ rtx: various improvements
+ Use locking
+ Don't push from the event handler, collected packets in a queue and push from
+ the chain function.
+ Clear queues on shutdown.
+
+2013-08-21 16:50:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ session: generate events correctly
+ Do correct shifting of the bitmask for lost packets.
+
+2013-08-21 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpmanager.c:
+ rtp: register rtx element better
+
+2013-08-21 16:32:50 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others
+ Probably fixes
+ https://bugzilla.gnome.org/show_bug.cgi?id=705477
+
+2013-08-21 13:03:34 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * ext/jpeg/gstjpegenc.c:
+ jpegenc: don't ignore return value from _finish_frame()
+ gst_video_encoder_finish_frame() will return FLOW_OK here if
+ there's no output buffer.
+
+2013-08-21 12:56:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpjpegdepay.c:
+ jpegdepay: add some more debug
+
+2013-08-21 12:10:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstdepay.h:
+ rtpgstdepay: only push events when they changed
+ Keep track of the STREAM_START and TAG events and only push them
+ when they changed.
+
+2013-08-21 10:52:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: taglists should not be merged in 1.0
+
+2013-08-21 10:28:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ rtpgstdepay: flush on FLUSH_STOP event
+
+2013-08-21 10:03:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: reset on state change
+ Do full reset on state change to READY
+
+2013-08-21 09:55:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: reset on FLUSH_STOP
+ Clear the adapter and pending buffer list on FLUSH_STOP.
+
+2013-08-21 09:39:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: don't use clock for config interval
+ We can't use the clock to time our config-interval because we are not
+ live (or there might not be a clock or the clock might not be running).
+ Instead just simply take the timestamp diff.
+
+2013-08-21 09:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.h:
+ rtpgstay: don't use // comments
+
+2013-08-08 11:55:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix response argument in handle-request signal
+
+2013-08-08 11:54:41 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Add sdes property and proxy it to rtpbin
+
+2013-08-07 09:47:35 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtpgstpay.h:
+ Send a stream-start whenever we send tags This is to make sure tags are cleared on the client if the stream-start was previously lost, otherwise, the client may end up with a merged taglist of multiple songs
+
+2013-07-25 21:12:05 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtpgstpay.h:
+ rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval This is useful in case the packet containing the inlined caps was lost or if new client joins an already running RTP stream and they missed the previous tag events. This also makes the payloader keep a list of merged tags so the retransmitted tag event contains all previously received. A STREAM_START event will flush the list of tags.
+
+2013-07-25 21:10:10 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time
+
+2013-07-25 21:03:34 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps
+
+2013-07-25 20:54:50 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtpgstpay.h:
+ rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList This is necessary to fix event/caps sending. If we send a STREAM_START packet, it will cause an error because the stream didn't receive its caps and new-segment events, so we must wait for the first buffer before sending the stream-start event buffer. However, the caps will be sent at the same time and so the 'inline caps' will be set for the event. We need to be able to payload individual packets (data, caps or events) and only send them when we call flush.
+
+2013-07-25 17:56:38 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START
+
+2013-07-25 17:52:16 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3
+
+2013-08-20 14:36:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: handle EOS
+ When the queue is empty, and we received EOS, pause and push an EOS
+ event downstream.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387
+
+2013-08-20 10:26:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: update docs
+
+2013-08-20 10:25:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: update all timers
+ Keep looping over all registered timers so that we can mark them lost instead of
+ stopping as soon as we find the timer for the current seqnum.
+
+2013-08-20 08:55:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: remove unused variables
+
+2013-08-19 21:10:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: reorganize timer handling
+ Restructure handling of incomming packet and the gap with the expected seqnum
+ and register all timers from the _chain function.
+ Convert a timer to a LOST packet timer when the max amount of retransmission
+ requests has been reached.
+
+2013-08-19 21:37:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: refactor packet spacing calculation
+
+2013-08-19 21:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: keep track of last seqnum and dts
+
+2013-08-19 21:29:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: small cleanups
+
+2013-08-19 21:21:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: reset retransmission timers in add/reschedule
+ Reset the retransmission timers when adding and rescheduling a timer.
+
+2013-08-19 21:12:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: rename variables for packet spacing
+
+2013-08-19 14:58:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: remove lost timer when we get the packet
+ When we receive a packet, also remove the LOST timer for it.
+
+2013-08-19 14:56:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: expected seqnum must increase
+ Only update the expected seqnum when it is bigger than the previous expected
+ seqnum.
+
+2013-08-19 14:55:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: add more debug
+
+2013-08-12 16:15:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/Makefile.am:
+ * gst/rtpmanager/gstrtpmanager.c:
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxqueue.h:
+ rtxqueue: add retransmission queue element
+
+2013-08-12 14:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ session: add some docs
+
+2013-08-06 16:29:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ session: handle NACK feedback and generate events
+ Handle and parse the feedback NACK packets and generate a Retransmission
+ event for each NACKed packet
+
+2013-08-19 13:19:42 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Add forward declaration for gst_v4l2_object_get_format_list
+
+2012-10-22 17:58:07 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/gstv4l2sink.h:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2: De-duplicate caps probing between src and sink
+
+2013-08-13 17:32:17 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/pulse/Makefile.am:
+ * ext/pulse/pulseprobe.c:
+ * ext/pulse/pulseprobe.h:
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesink.h:
+ * ext/pulse/pulsesrc.c:
+ * ext/pulse/pulsesrc.h:
+ pulse: Remove unused GstPulseProbe
+
+2013-08-19 12:46:45 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * sys/v4l2/gstv4l2tuner.c:
+ * sys/v4l2/tuner.c:
+ * sys/v4l2/tunerchannel.c:
+ * sys/v4l2/tunernorm.c:
+ v4l2: Use G_DEFINE_ macros for added thread safety
+
+2013-08-17 11:28:13 +0200 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/videomixer/videomixer2.c:
+ * gst/videomixer/videomixer2.h:
+ videomixer: Do not send flush_stop ourself after a flush_start
+ When we receive a flush_start, we should wait for the next flush_stop
+ and foward it, not create a flush_stop ourself.
+
+2013-08-16 17:10:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph264depay.c:
+ h264depay: init debug category early
+ Init the debug variable when we register the element because it is also used by
+ the payloader element when it calls the add_sps_pps method.
+
+2013-08-16 13:26:28 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: Properly set headers via the base class instead of just pushing them downstream
+ Prevents buffers from being send before the caps and segment events.
+
+2013-08-15 10:59:10 +0100 Chris Bass <floobleflam@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: check denominator isn't zero before scaling duration.
+ When gst_qtdemux_configure_stream sets fps_d, check that n_samples is
+ non-zero before using it as a denominator to scale the stream duration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=706076
+
+2013-08-15 15:08:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ext/jpeg/gstjpegdec.c:
+ * ext/jpeg/gstjpegenc.c:
+ * ext/libpng/gstpngdec.c:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ ext: Use new flush vfunc of video codec base classes and remove reset implementations
+
+2013-08-14 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: forward flush before stopping dataflow
+ First forward the flush event and then stop our loop function.
+
+2013-08-14 13:10:32 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ configure: require libsoup >= 2.38
+ Bump libsoup requirement for newer API used, like headers_get_one().
+ 2.38 is from early 2012 and is in linen with our GLib requirement.
+
+2013-08-14 11:54:19 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * ext/soup/gstsouphttpsrc.c:
+ soup: don't use deprecated soup_message_headers_get() API
+
+2013-08-13 17:44:50 +0200 Edward Hervey <edward@collabora.com>
+
+ * .gitignore:
+ .gitignore: Ignore files from automake test-driver
+
+2013-08-12 15:28:34 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ rtph264pay: Use the SPS/PPS handling function from the depayloader
+ Remove duplicated copies
+ https://bugzilla.gnome.org/show_bug.cgi?id=705553
+
+2013-08-12 15:26:08 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264depay.h:
+ rtph264depay: Make the SPS/PPS deduplication function generic
+ Make it not touch any internals of the depayloader
+ https://bugzilla.gnome.org/show_bug.cgi?id=705553
+
+2013-08-13 14:09:20 +0100 Chris Bass <floobleflam@gmail.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: allow conversion from raw AAC to ADTS
+ This patch will prepend ADTS headers to raw AAC audio frames, allowing
+ upstream elements to link to decoders that only support AAC in ADTS format.
+ Note that no error correction bits are added to ADTS frames in this code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=615740
+
+2013-08-13 12:44:11 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Only free GCheckSum after its last usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=705760
+
+2013-08-13 12:02:29 +0200 Andoni Morales Alastruey <ylatuya@gmail.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: fix critical setting a NULL uri redirection
+
+2013-07-13 01:50:56 +0200 Andoni Morales Alastruey <ylatuya@gmail.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: add redirection to the URI query
+
+2013-07-31 10:42:07 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: elst should offset samples instead of buffers
+ The current approach where buffers are offset is not ideal, as during seek
+ and loop current time is compared to sample times.
+ https://bugzilla.gnome.org/show_bug.cgi?id=700264
+
+2013-08-07 19:32:07 +0200 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/videomixer/videomixer2.c:
+ * tests/check/elements/videomixer.c:
+ videomixer: Send EOS if buf_end >= segment.stop
+ That means the whole segment is already played, and we are sure we
+ are EOS at that point.
+ Also handle segment seeks, and do not send EOS in that case.
+
+2013-08-04 14:40:38 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: send proper stream_start event
+ https://bugzilla.gnome.org//show_bug.cgi?id=705449
+
+2013-08-08 11:51:17 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/matroska/ebml-read.c:
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Don't print warnings during flushing and stop as soon as possible
+ https://bugzilla.gnome.org//show_bug.cgi?id=705442
+
+2013-08-07 11:14:38 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpvp8depay: mark key frames and delta frames properly
+ https://bugzilla.gnome.org/show_bug.cgi?id=705550
+
+2013-08-05 23:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ session: add NACK feedback in RTCP
+
+2013-08-05 23:22:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ source: add methods to register NACK
+ Add a method to register a missing packet for an ssrc along with
+ methods to get the missing packets and clear them.
+
+2013-08-04 23:05:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ session: handle Retransmission event and schedule NACK
+ Handle the retransmission event from downstream and use it to schedule a NACK
+ request.
+
+2013-08-05 23:20:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ session: pass data to remove func
+ Pass the data to the remove function because we are going to deref it when there
+ is pli or fir.
+
+2013-08-06 15:28:50 +0200 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix compilation
+
+2013-08-06 15:17:44 +0200 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE
+
+2013-08-06 11:58:38 +0200 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Make sure to send EOS if the buffer end time equals the segment end time
+ Otherwize EOS never gets sent in that particular case.
+
+2013-08-05 08:49:50 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
+
+ * gst/goom/gstgoom.c:
+ goom: Ensure src caps are writable
+ In some cases the src caps determined by goom weren't writable, causing
+ a bunch of assertion failures and failed caps. Fixed by always
+ explicitely making the caps writable
+ https://bugzilla.gnome.org/show_bug.cgi?id=705475
+
+2013-08-04 23:18:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ session: use common send_rtcp method
+ Reuse the send_rtcp method that already asks for the current time when
+ requesting a keyframe.
+
+2013-08-04 23:12:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ session: Don't use ClockTimeDiff for unsigned delays
+
+2013-08-04 16:52:15 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use buffer PTS if DTS is not set
+ Avoids ending up with completely bogus scaled duration/pts when new
+ buffers have invalid DTS.
+
+2013-08-04 14:32:47 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: skip https test if there's no TLS support in soup/glib
+
+2013-08-04 11:20:41 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp/gstrtpdec.c:
+ rtpdec: use generic marshaller
+
+2013-08-04 10:52:33 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * Makefile.am:
+ * sys/v4l2/.gitignore:
+ * sys/v4l2/Makefile.am:
+ * sys/v4l2/gstv4l2-marshal.list:
+ * sys/v4l2/tuner-marshal.list:
+ * sys/v4l2/tuner.c:
+ * sys/v4l2/tuner.h:
+ * win32/MANIFEST:
+ * win32/common/tuner-enumtypes.c:
+ * win32/common/tuner-enumtypes.h:
+ * win32/common/tuner-marshal.c:
+ * win32/common/tuner-marshal.h:
+ v4l2: remove unused enumtypes and use generic marshaller
+
+2013-08-04 10:47:38 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * Makefile.am:
+ * gst/udp/.gitignore:
+ * win32/common/gstudp-enumtypes.c:
+ * win32/common/gstudp-enumtypes.h:
+ * win32/common/gstudp-marshal.c:
+ * win32/common/gstudp-marshal.h:
+ udp: remove unused marshal and enumtypes files
+
+2013-08-04 09:38:19 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * Makefile.am:
+ * gst/rtpmanager/.gitignore:
+ * gst/rtpmanager/Makefile.am:
+ * gst/rtpmanager/gstrtpbin-marshal.list:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpptdemux.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/rtpmanager/rtpsession.c:
+ * win32/MANIFEST:
+ * win32/common/gstrtpbin-marshal.c:
+ * win32/common/gstrtpbin-marshal.h:
+ rtpmanager: use generic marshaller
+
+2013-08-04 00:13:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: send event in right direction
+
+2013-08-02 17:38:34 -0700 David Schleef <ds@schleef.org>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ tests: create/remove orc directory at proper time
+ Before automake creates .deps directories, and during distclean.
+
+2013-08-03 00:25:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ session: add FIR and PLI like other RTCP packets
+ Add the FIR and PLI packets like the other RTCP packet instead of from the
+ on-sending-rtcp default signal handler.
+
+2013-08-02 17:22:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: fix property ranges
+
+2013-08-02 16:42:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: push retransmission events
+
+2013-08-02 14:12:16 +0200 Lubosz Sarnecki <lubosz@gmail.com>
+
+ * configure.ac:
+ build: add subdir-objects to AM_INIT_AUTOMAKE
+ Fixes warnings with automake 1.14
+ https://bugzilla.gnome.org/show_bug.cgi?id=705350
+
+2013-08-02 14:54:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: add support for retransmission retry
+ When we didn't receive a packet after requesting retransmission, retry
+ asking for retransmission for a certain period.
+
+2013-08-02 14:19:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: add properties
+ Add properties to control retransmission parameters
+
+2013-08-02 12:44:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: use corrected timeout when rescheduling
+ When we recalculate the timeout, use the corrected timeout value depending on
+ the timer type.
+
+2013-08-02 12:43:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: update timers after queueing
+ Else we might update the timer needlessly for duplicates.
+
+2013-08-02 12:42:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: move method up
+
+2013-08-02 06:28:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: small cleanup
+
+2013-08-01 23:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: unschedule old expected packets
+ When we receive a new packet, unschedule old outstanding packets when their
+ seqnum is too far away.
+
+2013-08-01 23:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: refactor timer update
+
+2013-08-01 23:24:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: update timers when removing
+ Update the timers when we remove a timer.
+ Handle canceled timers, make them unschedule the current timer and
+ trigger the timeout code.
+
+2013-08-01 23:22:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: fix typo
+
+2013-08-01 15:40:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: improve timeout management
+ If we change the seqnum of an existing timer and we were waiting for
+ that timer, unschedule it. If we change the timeout of an existing timer and we
+ were waiting on it, only unschedule when the new time is smaller.
+
+2013-08-01 15:05:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: install timer for expected arrival
+ Install a timer that is triggered when the expected arrival time of a packet
+ expired.
+
+2013-08-01 14:56:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: improve unschedule of timers
+ Conflicts:
+ gst/rtpmanager/gstrtpjitterbuffer.c
+
+2013-08-01 12:21:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: move code around
+
+2013-08-01 12:07:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: estimate inter packet spacing
+ When we see two packets with consecutive seqnums and a different RTP time, use
+ the DTS difference as the inter packet spacing estimate.
+
+2013-08-01 12:01:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: keep track of current timeout
+
+2013-08-01 11:49:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: cleanup timer handling
+
+2013-08-01 11:40:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: reset is only possible with a GAP
+
+2013-08-01 11:29:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ jitterbuffer: operate on DTS
+ Make the jitterbuffer schedule the timeouts based on the DTS instead
+ of the PTS. This makes it all smoother with reordered frames and gives
+ the decoder time to reorder the frames in time.
+
+2013-08-01 11:14:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: rename timout variable
+
+2013-07-31 17:08:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: small cleanup
+
+2013-07-31 16:59:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: block output in paused or buffering
+
+2013-07-31 16:59:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: store pts in timer
+ Only store the pts in the timer so that we can both do timeouts with timings on
+ the input and output of the jitterbuffer.
+
+2013-07-30 23:14:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: refactor jitterbuffer
+ Refactor the jitterbuffer code. Make separate function for peeking a buffer,
+ pushing the next buffer, waiting for timeouts and handling the timeouts.
+ The main loop now tries to push as many buffers as it can until it runs out of
+ buffers or when it detects a seqnum discont. Then it will wait for some event to
+ happen before attempting to push more buffers.
+ Make methods to register timeouts in an array. These timeouts are registered
+ when we detect a missing packet, sync for the first packet or when we find an
+ estimation for the end-of-stream.
+ This greatly simplifies and clarifies the code and also makes it possible to
+ register more complicated timeout schemes later.
+
+2013-07-30 18:52:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: use NULL to ignore percent
+ If we pass NULL to pop and push we ignore the percent result.
+
+2013-07-30 07:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: refactor
+ Move eos estimation into separate function
+
+2013-07-30 14:28:19 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: don't leak stream_id string
+ https://bugzilla.gnome.org/show_bug.cgi?id=705142
+
+2013-07-29 19:53:52 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * po/LINGUAS:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/ja.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sl.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translations
+
+2013-07-29 19:48:54 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/check/elements/.gitignore:
+ tests: ignore new test binaries
+
+2013-07-29 14:47:49 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ Back to development
+
=== release 1.1.3 ===
-2013-07-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2013-07-29 13:42:18 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.1.3
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * 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-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.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-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.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-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.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-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.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 1.1.3
+
+2013-07-29 13:42:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.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/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.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:
+ Update .po files
2013-07-29 12:12:41 +0200 Sebastian Dröge <slomo@circular-chaos.org>
-This is GStreamer Good Plugins 1.1.3
+This is GStreamer Good Plugins 1.1.4
-Release notes for GStreamer Good Plugins 1.1.3
+Release notes for GStreamer Good Plugins 1.1.4
The GStreamer team is proud to announce a new bug-fix release
-Features of this release
-
-
Bugs fixed in this release
- * 672227 : qtdemux: need mapping for video/x-gst-fourcc-xdvc
- * 674626 : [rtpsession] High CPU usage in rtcp thread when setting RR bandwidth = 0
- * 696228 : mp4mux: streamable=true seeks on sink at EOS, causing errors with some sinks
- * 699791 : qtdemux: certain videos from Vivotek camera do not play - " This file is corrupt and cannot be played "
+ * 705371 : souphttpsrc: Does network operations from the state change thread
+ * 590768 : GstPulseSrc should allow swapping the device used by the stream
+ * 637754 : multipartdemux: time stamp output buffer based on first input buffer not last
+ * 694445 : pulsesink: add support for AAC pass-through
* 700264 : qtdemux: ignores first editlist
- * 700933 : avidemux: Wrong duration in push mode for some clips.
- * 703560 : [PATCH] rtpbin: new event for informing downstream about synchronization
- * 703901 : audioecho: Output broken if delay value got changed when playing or paused
- * 704007 : qtdemux: skips disabled tracks
- * 704053 : souphttpsrc: Don't treat errors on HEAD request as fatal
- * 704193 : avidemux: no PTS nor DTS on decoded frame
- * 704241 : souphttpsrc: Ignore all HEAD errors
- * 704255 : qtdemux: invalid base time on newsegment in push mode after a seek
- * 704272 : qtdemux: invalid seek offset used because of chapters
- * 704292 : qtdemux: does not know WRAW fourcc
- * 704294 : qtdemux: videos in Windows WRLE format don't work
- * 704357 : regression: videocrop unit test fail after adding GRAY16_BE format
- * 704547 : vpx: fix compilation when decoder or encoder package is not build
- * 704548 : qtdemux: set proper caps for mpeg-1 audio
- * 704921 : matroskademux: memory leak in check_subtitle_buffer
- * 704951 : avidemux: memory leak
- * 704996 : qtdemux: support for H264 fourcc
+ * 702988 : gdkpixbufoverlay: crashes if any property changes during playback when location property is not set
+ * 705108 : rtpgstpay: Add a config-interval property
+ * 705442 : matroskademux: prints warnings on seek
+ * 705449 : avidemux: sends stream_start event without group_id
+ * 705553 : rtph264pay: Entries of SPS and PPS duplicated
+ * 705760 : rtspsrc produces GLib error
+ * 705959 : souphttpsrc deprecated symbols
+ * 706076 : qtdemux: failed assertion for fragmented mp4 (DASH) streams
+ * 706642 : avimux: unmap the correct buffer
==== Download ====
Contributors to this release
- * Alessandro Decina
+ * Akihiro Tsukada
* Andoni Morales Alastruey
- * Arnaud Vrac
- * Carlos Rafael Giani
- * Douglas Bagnall
+ * Chris Bass
+ * David Schleef
* Edward Hervey
- * Kyosuke Nekomura
+ * Kishore Arepalli
+ * Lubosz Sarnecki
* Matej Knopp
* Mathieu Duponchelle
- * Nicolas Dufresne
+ * Michael Olbrich
* Olivier Crête
* Sebastian Dröge
- * Thiago Santos
+ * Sjoerd Simons
+ * Thibault Saunier
* Tim-Philipp Müller
- * Vincent Penquerc'h
* Wim Taymans
+ * Youness Alaoui
\ No newline at end of file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/pre
-AC_INIT([GStreamer Good Plug-ins],[1.1.3.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
+AC_INIT([GStreamer Good Plug-ins],[1.1.4],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AG_GST_INIT
[GStreamer API Version])
AG_GST_LIBTOOL_PREPARE
-AS_LIBTOOL(GST, 103, 0, 103)
+AS_LIBTOOL(GST, 104, 0, 104)
dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.1.3.1
-GSTPB_REQ=1.1.3.1
+GST_REQ=1.1.4
+GSTPB_REQ=1.1.4
dnl *** autotools stuff ****
</ARG>
<ARG>
+<NAME>GstRTSPSrc::sdes</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>SDES</NICK>
+<BLURB>The SDES items of this session.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstRTPDec::skip</NAME>
<TYPE>gint</TYPE>
<RANGE></RANGE>
</ARG>
<ARG>
+<NAME>GstPulseSink::current-device</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Current Device</NICK>
+<BLURB>The current PulseAudio sink device.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstPulseSrc::device</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
</ARG>
<ARG>
+<NAME>GstPulseSrc::current-device</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Current Device</NICK>
+<BLURB>The current PulseAudio source device.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstPulseMixer::device</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
</ARG>
<ARG>
+<NAME>GstRtpBin::do-retransmission</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Do retransmission</NICK>
+<BLURB>Send an event downstream to request packet retransmission.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstRtpJitterBuffer::do-lost</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
</ARG>
<ARG>
+<NAME>GstRtpJitterBuffer::do-retransmission</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Do Retransmission</NICK>
+<BLURB>Send retransmission events upstream when a packet is late.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-delay</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>r</FLAGS>
+<NICK>RTX Delay</NICK>
+<BLURB>Extra time in ms to wait before sending retransmission event (-1 automatic).</BLURB>
+<DEFAULT>20</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-delay-reorder</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>r</FLAGS>
+<NICK>RTX Delay Reorder</NICK>
+<BLURB>Sending retransmission event when this much reordering (-1 automatic).</BLURB>
+<DEFAULT>3</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-retry-period</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>r</FLAGS>
+<NICK>RTX Retry Period</NICK>
+<BLURB>Try to get a retransmission for this many ms (-1 automatic).</BLURB>
+<DEFAULT>160</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-retry-timeout</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>r</FLAGS>
+<NICK>RTX Retry Timeout</NICK>
+<BLURB>Retry sending a transmission event after this timeout in ms (-1 automatic).</BLURB>
+<DEFAULT>40</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstRtpSession::bandwidth</NAME>
<TYPE>gdouble</TYPE>
<RANGE>>= 0</RANGE>
<DEFAULT>0</DEFAULT>
</ARG>
+<ARG>
+<NAME>GstRTPRtxQueue::max-size-packets</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max Size Packets</NICK>
+<BLURB>Amount of packets to queue (0 = unlimited).</BLURB>
+<DEFAULT>100</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPRtxQueue::max-size-time</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max Size Times</NICK>
+<BLURB>Amount of ms to queue (0 = unlimited).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpGSTPay::config-interval</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 3600</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Caps/Tags Send Interval</NICK>
+<BLURB>Interval for sending caps and TAG events in seconds (0 = disabled).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
GstRTPDec
GstRTPMux
GstRTPDTMFMux
+ GstRTPRtxQueue
GstRndBufferSize
GstRtpJitterBuffer
GstRtpPtDemux
<description>Source for video data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ASCII Art video sink</description>
<filename>../../ext/aalib/.libs/libgstaasink.so</filename>
<basename>libgstaasink.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ALaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstalaw.so</filename>
<basename>libgstalaw.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>adds an alpha channel to video - constant or via chroma-keying</description>
<filename>../../gst/alpha/.libs/libgstalpha.so</filename>
<basename>libgstalpha.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>RGBA from/to AYUV colorspace conversion preserving the alpha channel</description>
<filename>../../gst/alpha/.libs/libgstalphacolor.so</filename>
<basename>libgstalphacolor.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>APEv1/2 tag reader</description>
<filename>../../gst/apetag/.libs/libgstapetag.so</filename>
<basename>libgstapetag.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio effects plugin</description>
<filename>../../gst/audiofx/.libs/libgstaudiofx.so</filename>
<basename>libgstaudiofx.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Parsers for various audio formats</description>
<filename>../../gst/audioparsers/.libs/libgstaudioparsers.so</filename>
<basename>libgstaudioparsers.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>parses au streams</description>
<filename>../../gst/auparse/.libs/libgstauparse.so</filename>
<basename>libgstauparse.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Plugin contains auto-detection plugins for video/audio in- and outputs</description>
<filename>../../gst/autodetect/.libs/libgstautodetect.so</filename>
<basename>libgstautodetect.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>AVI stream handling</description>
<filename>../../gst/avi/.libs/libgstavi.so</filename>
<basename>libgstavi.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Colored ASCII Art video sink</description>
<filename>../../ext/libcaca/.libs/libgstcacasink.so</filename>
<basename>libgstcacasink.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Cairo-based elements</description>
<filename>../../ext/cairo/.libs/libgstcairo.so</filename>
<basename>libgstcairo.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio Cutter to split audio into non-silent bits</description>
<filename>../../gst/cutter/.libs/libgstcutter.so</filename>
<basename>libgstcutter.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>elements for testing and debugging</description>
<filename>../../gst/debugutils/.libs/libgstdebug.so</filename>
<basename>libgstdebug.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Deinterlacer</description>
<filename>../../gst/deinterlace/.libs/libgstdeinterlace.so</filename>
<basename>libgstdeinterlace.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>DTMF plugins</description>
<filename>../../gst/dtmf/.libs/libgstdtmf.so</filename>
<basename>libgstdtmf.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
<filename>../../ext/dv/.libs/libgstdv.so</filename>
<basename>libgstdv.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>effect plugins from the effectv project</description>
<filename>../../gst/effectv/.libs/libgsteffectv.so</filename>
<basename>libgsteffectv.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GStreamer audio equalizers</description>
<filename>../../gst/equalizer/.libs/libgstequalizer.so</filename>
<basename>libgstequalizer.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>The FLAC Lossless compressor Codec</description>
<filename>../../ext/flac/.libs/libgstflac.so</filename>
<basename>libgstflac.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>FLV muxing and demuxing plugin</description>
<filename>../../gst/flv/.libs/libgstflv.so</filename>
<basename>libgstflv.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>FLC/FLI/FLX video decoder</description>
<filename>../../gst/flx/.libs/libgstflxdec.so</filename>
<basename>libgstflxdec.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GdkPixbuf-based image decoder, overlay and sink</description>
<filename>../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so</filename>
<basename>libgstgdkpixbuf.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GOOM visualization filter</description>
<filename>../../gst/goom/.libs/libgstgoom.so</filename>
<basename>libgstgoom.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GOOM 2k1 visualization filter</description>
<filename>../../gst/goom2k1/.libs/libgstgoom2k1.so</filename>
<basename>libgstgoom2k1.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Demux ICY tags from a stream</description>
<filename>../../gst/icydemux/.libs/libgsticydemux.so</filename>
<basename>libgsticydemux.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Demux ID3v1 and ID3v2 tags from a file</description>
<filename>../../gst/id3demux/.libs/libgstid3demux.so</filename>
<basename>libgstid3demux.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Still frame stream generator</description>
<filename>../../gst/imagefreeze/.libs/libgstimagefreeze.so</filename>
<basename>libgstimagefreeze.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio interleaver/deinterleaver</description>
<filename>../../gst/interleave/.libs/libgstinterleave.so</filename>
<basename>libgstinterleave.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ISO base media file format support (mp4, 3gpp, qt, mj2)</description>
<filename>../../gst/isomp4/.libs/libgstisomp4.so</filename>
<basename>libgstisomp4.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>JACK audio elements</description>
<filename>../../ext/jack/.libs/libgstjack.so</filename>
<basename>libgstjack.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>JPeg plugin library</description>
<filename>../../ext/jpeg/.libs/libgstjpeg.so</filename>
<basename>libgstjpeg.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio level plugin</description>
<filename>../../gst/level/.libs/libgstlevel.so</filename>
<basename>libgstlevel.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Matroska and WebM stream handling</description>
<filename>../../gst/matroska/.libs/libgstmatroska.so</filename>
<basename>libgstmatroska.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<name>video_%u</name>
<direction>sink</direction>
<presence>request</presence>
- <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-huffyuv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-dv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-msmpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-theora; video/x-dirac, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-pn-realvideo, rmversion=(int)[ 1, 4 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-vp8, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw, format=(string){ YUY2, I420, YV12, UYVY, AYUV }, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-wmv, wmvversion=(int)[ 1, 3 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-huffyuv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-theora; video/x-dirac, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-pn-realvideo, rmversion=(int)[ 1, 4 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp8, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw, format=(string){ YUY2, I420, YV12, UYVY, AYUV }, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)[ 1, 3 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
</caps>
<caps>
<name>src</name>
<description>MuLaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstmulaw.so</filename>
<basename>libgstmulaw.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Reads/Writes buffers from/to sequentially named files</description>
<filename>../../gst/multifile/.libs/libgstmultifile.so</filename>
<basename>libgstmultifile.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>multipart stream manipulation</description>
<filename>../../gst/multipart/.libs/libgstmultipart.so</filename>
<basename>libgstmultipart.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Template for a video filter</description>
<filename>../../gst/debugutils/.libs/libgstnavigationtest.so</filename>
<basename>libgstnavigationtest.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Open Sound System (OSS) version 4 support for GStreamer</description>
<filename>../../sys/oss4/.libs/libgstoss4audio.so</filename>
<basename>libgstoss4audio.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>OSS (Open Sound System) support for GStreamer</description>
<filename>../../sys/oss/.libs/libgstossaudio.so</filename>
<basename>libgstossaudio.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>PNG plugin library</description>
<filename>../../ext/libpng/.libs/libgstpng.so</filename>
<basename>libgstpng.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>PulseAudio plugin library</description>
<filename>../../ext/pulse/.libs/libgstpulse.so</filename>
<basename>libgstpulse.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
- <details>audio/x-raw, format=(string){ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, S24LE, S24BE, S24_32LE, S24_32BE, U8 }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-alaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-mulaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-ac3, framed=(boolean)true; audio/x-eac3, framed=(boolean)true; audio/x-dts, framed=(boolean)true, block-size=(int){ 512, 1024, 2048 }; audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)[ 1, 2 ], parsed=(boolean)true</details>
+ <details>audio/x-raw, format=(string){ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, S24LE, S24BE, S24_32LE, S24_32BE, U8 }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-alaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-mulaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-ac3, framed=(boolean)true; audio/x-eac3, framed=(boolean)true; audio/x-dts, framed=(boolean)true, block-size=(int){ 512, 1024, 2048 }; audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)[ 1, 2 ], parsed=(boolean)true; audio/mpeg, mpegversion=(int){ 2, 4 }, framed=(boolean)true, stream-format=(string)adts</details>
</caps>
</pads>
</element>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
- <details>audio/x-raw, format=(string){ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, U8 }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-alaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-mulaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]</details>
+ <details>audio/x-raw, format=(string){ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, S24LE, S24BE, S24_32LE, S24_32BE, U8 }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-alaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]; audio/x-mulaw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 32 ]</details>
</caps>
</pads>
</element>
<description>ReplayGain volume normalization</description>
<filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
<basename>libgstreplaygain.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Real-time protocol plugins</description>
<filename>../../gst/rtp/.libs/libgstrtp.so</filename>
<basename>libgstrtp.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
</pads>
</element>
<element>
+ <name>rtprtxqueue</name>
+ <longname>RTP Retransmission Queue</longname>
+ <class>Codec</class>
+ <description>Keep RTP packets in a queue for retransmission</description>
+ <author>Wim Taymans <wim.taymans@gmail.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>application/x-rtp</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>application/x-rtp</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
<name>rtpsession</name>
<longname>RTP Session</longname>
<class>Filter/Network/RTP</class>
<description>transfer data via RTSP</description>
<filename>../../gst/rtsp/.libs/libgstrtsp.so</filename>
<basename>libgstrtsp.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Shape Wipe transition filter</description>
<filename>../../gst/shapewipe/.libs/libgstshapewipe.so</filename>
<basename>libgstshapewipe.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Sends data to an icecast server using libshout2</description>
<filename>../../ext/shout2/.libs/libgstshout2.so</filename>
<basename>libgstshout2.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>libshout2</package>
<description>Apply the standard SMPTE transitions on video images</description>
<filename>../../gst/smpte/.libs/libgstsmpte.so</filename>
<basename>libgstsmpte.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>libsoup HTTP client src/sink</description>
<filename>../../ext/soup/.libs/libgstsouphttpsrc.so</filename>
<basename>libgstsouphttpsrc.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Run an FFT on the audio signal, output spectrum data</description>
<filename>../../gst/spectrum/.libs/libgstspectrum.so</filename>
<basename>libgstspectrum.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Speex plugin library</description>
<filename>../../ext/speex/.libs/libgstspeex.so</filename>
<basename>libgstspeex.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Tag writing plug-in based on taglib</description>
<filename>../../ext/taglib/.libs/libgsttaglib.so</filename>
<basename>libgsttaglib.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>transfer data via UDP</description>
<filename>../../gst/udp/.libs/libgstudp.so</filename>
<basename>libgstudp.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>elements for Video 4 Linux</description>
<filename>../../sys/v4l2/.libs/libgstvideo4linux2.so</filename>
<basename>libgstvideo4linux2.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>resizes a video by adding borders or cropping</description>
<filename>../../gst/videobox/.libs/libgstvideobox.so</filename>
<basename>libgstvideobox.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Crops video into a user-defined region</description>
<filename>../../gst/videocrop/.libs/libgstvideocrop.so</filename>
<basename>libgstvideocrop.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Video filters plugin</description>
<filename>../../gst/videofilter/.libs/libgstvideofilter.so</filename>
<basename>libgstvideofilter.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Video mixer</description>
<filename>../../gst/videomixer/.libs/libgstvideomixer.so</filename>
<basename>libgstvideomixer.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>VP8 plugin</description>
<filename>../../ext/vpx/.libs/libgstvpx.so</filename>
<basename>libgstvpx.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Encode raw audio into WAV</description>
<filename>../../gst/wavenc/.libs/libgstwavenc.so</filename>
<basename>libgstwavenc.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Parse a .wav file into raw audio</description>
<filename>../../gst/wavparse/.libs/libgstwavparse.so</filename>
<basename>libgstwavparse.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>X11 video input plugin using standard Xlib calls</description>
<filename>../../sys/ximage/.libs/libgstximagesrc.so</filename>
<basename>libgstximagesrc.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Encodes a YUV frame into the yuv4mpeg format (mjpegtools)</description>
<filename>../../gst/y4m/.libs/libgsty4menc.so</filename>
<basename>libgsty4menc.so</basename>
- <version>1.1.3</version>
+ <version>1.1.4</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<release>
<Version>
+ <revision>1.1.4</revision>
+ <branch>1.1</branch>
+ <name></name>
+ <created>2013-08-28</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.1.4.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.1.3</revision>
<branch>1.1</branch>
<name></name>
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 110, 111, 110, 101, 11, 4, 4, 12, 2, 2,
195, 0, 4, 4, 2, 0,
};
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 110, 111, 110, 101, 11, 8, 8, 12, 4, 4,
194, 0, 4, 4, 2, 0,
};
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 110, 111, 110, 101, 11, 4, 4, 12, 4, 4,
21, 1, 79, 0, 4, 2, 0,
};
static const orc_uint8 bc[] = {
1, 9, 39, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 110, 111, 110, 101, 11, 8, 8, 12, 8, 8,
21, 1, 112, 0, 4, 2, 0,
};
static const orc_uint8 bc[] = {
1, 9, 38, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 112, 115, 121, 11, 4, 4, 12, 2, 2, 17,
4, 17, 4, 20, 8, 20, 4, 20, 4, 153, 33, 4, 211, 33, 33, 202,
34, 33, 25, 202, 33, 33, 24, 194, 32, 33, 34, 21, 1, 210, 32, 32,
static const orc_uint8 bc[] = {
1, 9, 38, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 112, 115, 121, 11, 8, 8, 12, 4, 4, 17,
4, 17, 4, 20, 4, 20, 4, 202, 33, 4, 25, 202, 32, 4, 24, 194,
0, 32, 33, 2, 0,
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 112, 115, 121, 95, 114, 105, 103, 104, 116, 11,
4, 4, 12, 4, 4, 17, 4, 17, 4, 20, 8, 20, 4, 20, 4, 20,
4, 21, 1, 153, 32, 4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 34,
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 112, 115, 121, 95, 108, 101, 102, 116, 11, 4,
4, 12, 4, 4, 17, 4, 17, 4, 20, 8, 20, 4, 20, 4, 20, 4,
21, 1, 153, 32, 4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 35, 32,
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 112, 115, 121, 95, 114, 105, 103, 104, 116, 11,
8, 8, 12, 8, 8, 17, 4, 17, 4, 20, 4, 20, 4, 20, 4, 192,
32, 4, 193, 33, 4, 202, 34, 32, 25, 202, 32, 32, 24, 200, 33, 34,
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 112, 115, 121, 95, 108, 101, 102, 116, 11, 8,
8, 12, 8, 8, 17, 4, 17, 4, 20, 4, 20, 4, 20, 4, 192, 32,
4, 193, 34, 4, 202, 33, 34, 24, 202, 34, 34, 25, 200, 32, 33, 32,
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
4, 4, 12, 2, 2, 17, 4, 20, 8, 20, 4, 20, 4, 153, 33, 4,
211, 33, 33, 202, 34, 33, 24, 194, 32, 33, 34, 21, 1, 210, 32, 32,
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 49, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 4,
4, 12, 2, 2, 17, 4, 20, 8, 20, 4, 20, 4, 153, 34, 4, 211,
34, 34, 202, 33, 34, 24, 194, 32, 33, 34, 21, 1, 210, 32, 32, 21,
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
4, 4, 12, 4, 4, 17, 4, 20, 8, 20, 4, 20, 4, 21, 1, 153,
32, 4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 34, 32, 202, 34, 34,
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 115,
- 49,
+ 49,
54, 95, 99, 104, 50, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 4,
4, 12, 4, 4, 17, 4, 20, 8, 20, 4, 20, 4, 21, 1, 153, 32,
4, 21, 1, 211, 32, 32, 192, 33, 32, 193, 34, 32, 202, 33, 33, 24,
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
8, 8, 12, 4, 4, 17, 4, 20, 4, 20, 4, 112, 32, 4, 202, 33,
4, 24, 194, 0, 32, 33, 2, 0,
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 49, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 8,
8, 12, 4, 4, 17, 4, 20, 4, 20, 4, 202, 32, 4, 24, 112, 33,
4, 194, 0, 32, 33, 2, 0,
static const orc_uint8 bc[] = {
1, 9, 44, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 115, 105, 109, 95, 114, 105, 103, 104, 116, 11,
8, 8, 12, 8, 8, 17, 4, 20, 4, 20, 4, 192, 32, 4, 193, 33,
4, 202, 33, 33, 24, 194, 0, 32, 33, 2, 0,
static const orc_uint8 bc[] = {
1, 9, 43, 97, 117, 100, 105, 111, 112, 97, 110, 111, 114, 97, 109, 97,
109, 95, 111, 114, 99, 95, 112, 114, 111, 99, 101, 115, 115, 95, 102,
- 51,
+ 51,
50, 95, 99, 104, 50, 95, 115, 105, 109, 95, 108, 101, 102, 116, 11, 8,
8, 12, 8, 8, 17, 4, 20, 4, 20, 4, 192, 32, 4, 193, 33, 4,
202, 32, 32, 24, 194, 0, 32, 33, 2, 0,
#define GST_PACKAGE_ORIGIN "Unknown package origin"
/* GStreamer package release date/time for plugins as YYYY-MM-DD */
-#define GST_PACKAGE_RELEASE_DATETIME "2013-07-29"
+#define GST_PACKAGE_RELEASE_DATETIME "2013-08-28"
/* Define if static plugins should be built */
#undef GST_PLUGIN_BUILD_STATIC
#define PACKAGE_NAME "GStreamer Good Plug-ins"
/* Define to the full name and version of this package. */
-#define PACKAGE_STRING "GStreamer Good Plug-ins 1.1.3"
+#define PACKAGE_STRING "GStreamer Good Plug-ins 1.1.4"
/* Define to the one symbol short name of this package. */
#define PACKAGE_TARNAME "gst-plugins-good"
#undef PACKAGE_URL
/* Define to the version of this package. */
-#define PACKAGE_VERSION "1.1.3"
+#define PACKAGE_VERSION "1.1.4"
/* directory where plugins are located */
#ifdef _DEBUG
#undef TARGET_CPU
/* Version number of package */
-#define VERSION "1.1.3"
+#define VERSION "1.1.4"
/* old wavpack API */
#undef WAVPACK_OLD_API
projects / platform / upstream / gst-plugins-good.git / commitdiff