ChangeLog
=========
2013-08-28 Sebastian Dröge <[email protected]>
* configure.ac:
releasing 1.1.4
2013-08-28 12:32:10 +0200 Sebastian Dröge <[email protected]>
* po/pt_BR.po:
po: update translations
2013-08-27 15:25:16 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/rtpsession.c:
session: add more debug
2013-08-27 09:34:46 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpsession.c:
jitterbuffer: fix types of the retransmission event
2013-08-27 09:33:03 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* configure.ac:
configure.ac: Don't set BZ2_LIBS if bz2 is not found
2013-08-26 11:50:27 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/rtpsession.c:
rtpsession: add some more debug
2013-08-20 22:12:03 +0200 Mathieu Duponchelle <[email protected]>
* 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 <[email protected]>
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartdemux.h:
multipartdemux: propagate discont
2013-08-23 15:49:47 +0100 Tim-Philipp Müller <[email protected]>
* gst/multipart/multipartdemux.c:
multipartdemux: remove dynamic sourcpads when going from PAUSED to
READY
2013-08-23 15:29:28 +0100 Tim-Philipp Müller <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtprtxqueue.c:
* gst/rtpmanager/gstrtprtxqueue.h:
rtxqueue: add property to configure queue size
2013-08-23 12:07:55 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
rtpbin: proxy jitterbuffer do-retransmission property
2013-08-23 11:17:45 +0200 Michael Olbrich <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpmanager.c:
rtp: register rtx element better
2013-08-21 16:32:50 +0200 Sebastian Dröge <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtp/gstrtpjpegdepay.c:
jpegdepay: add some more debug
2013-08-21 12:10:00 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtp/gstrtpgstpay.c:
rtpgstpay: taglists should not be merged in 1.0
2013-08-21 10:28:50 +0200 Wim Taymans <[email protected]>
* gst/rtp/gstrtpgstdepay.c:
rtpgstdepay: flush on FLUSH_STOP event
2013-08-21 10:03:52 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtp/gstrtpgstpay.h:
rtpgstay: don't use // comments
2013-08-08 11:55:22 -0400 Youness Alaoui <[email protected]>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Fix response argument in handle-request signal
2013-08-08 11:54:41 -0400 Youness Alaoui <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtp/gstrtpgstpay.c:
rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3
2013-08-20 14:36:59 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: update docs
2013-08-20 10:25:17 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: remove unused variables
2013-08-19 21:10:00 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: refactor packet spacing calculation
2013-08-19 21:34:38 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: keep track of last seqnum and dts
2013-08-19 21:29:49 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: small cleanups
2013-08-19 21:21:08 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: rename variables for packet spacing
2013-08-19 14:58:01 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: add more debug
2013-08-12 16:15:54 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/rtpsession.c:
session: add some docs
2013-08-06 16:29:54 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* sys/v4l2/gstv4l2object.c:
v4l2: Add forward declaration for gst_v4l2_object_get_format_list
2012-10-22 17:58:07 -0400 Olivier Crête <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* ext/soup/gstsouphttpsrc.c:
soup: don't use deprecated soup_message_headers_get() API
2013-08-13 17:44:50 +0200 Edward Hervey <[email protected]>
* .gitignore:
.gitignore: Ignore files from automake test-driver
2013-08-12 15:28:34 -0400 Olivier Crête <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: fix critical setting a NULL uri redirection
2013-07-13 01:50:56 +0200 Andoni Morales Alastruey <[email protected]>
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
souphttpsrc: add redirection to the URI query
2013-07-31 10:42:07 +0200 Matej Knopp <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/rtpsession.c:
session: add NACK feedback in RTCP
2013-08-05 23:22:16 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Fix compilation
2013-08-06 15:17:44 +0200 Thibault Saunier <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE
2013-08-06 11:58:38 +0200 Thibault Saunier <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtsp/gstrtpdec.c:
rtpdec: use generic marshaller
2013-08-04 10:52:33 +0100 Tim-Philipp Müller <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: send event in right direction
2013-08-02 17:38:34 -0700 David Schleef <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: fix property ranges
2013-08-02 16:42:52 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: push retransmission events
2013-08-02 14:12:16 +0200 Lubosz Sarnecki <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: add properties
Add properties to control retransmission parameters
2013-08-02 12:44:58 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: move method up
2013-08-02 06:28:32 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: small cleanup
2013-08-01 23:26:06 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: refactor timer update
2013-08-01 23:24:29 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: fix typo
2013-08-01 15:40:52 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: improve unschedule of timers
Conflicts:
gst/rtpmanager/gstrtpjitterbuffer.c
2013-08-01 12:21:53 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: move code around
2013-08-01 12:07:11 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: keep track of current timeout
2013-08-01 11:49:10 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: cleanup timer handling
2013-08-01 11:40:41 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: reset is only possible with a GAP
2013-08-01 11:29:32 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: rename timout variable
2013-07-31 17:08:58 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: small cleanup
2013-07-31 16:59:58 +0200 Wim Taymans <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: block output in paused or buffering
2013-07-31 16:59:09 +0200 Wim Taymans <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: refactor
Move eos estimation into separate function
2013-07-30 14:28:19 +0100 Tim-Philipp Müller <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* tests/check/elements/.gitignore:
tests: ignore new test binaries
2013-07-29 14:47:49 +0200 Sebastian Dröge <[email protected]>
* 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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