gst-plugins-good 1.1.4
- From: Tim-Philipp Müller <install-module master gnome org>
- To: FTP Releases <ftp-release-list gnome org>
- Subject: gst-plugins-good 1.1.4
- Date: Fri, 30 Aug 2013 22:25:14 +0000 (UTC)
ChangeLog
=========
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
Download
========
http://download.gnome.org/sources/gst-plugins-good/1.1/gst-plugins-good-1.1.4.tar.xz (2.71M)
sha256sum: 34728258775e152dbe8a25034cda91f2461abfa43ec0eef8aea06a87c4215df4
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