Commit 76293efd authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.1.4

parent 4bc1a78f
=== 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