Wim Taymans [Fri, 14 Sep 2012 14:45:34 +0000 (16:45 +0200)]
scaletempo: ffmpegcolorspace is no more
Sebastian Dröge [Thu, 5 Apr 2012 16:02:56 +0000 (18:02 +0200)]
scaletempo: Update for GST_PLUGIN_DEFINE() API changes
Mark Nauwelaerts [Sun, 18 Mar 2012 17:32:55 +0000 (18:32 +0100)]
scaletempo: port to 0.11
Stefan Kost [Thu, 7 Jul 2011 17:52:50 +0000 (10:52 -0700)]
scaletempo: improve the docs
Fix the syntax, add more explanation and xref the properties.
Chris E Jones [Tue, 22 Mar 2011 12:46:42 +0000 (13:46 +0100)]
scaletempo: Correctly handle newsegment events with stop==-1
Fixes bug #645420.
Stefan Kost [Tue, 19 Oct 2010 10:43:14 +0000 (13:43 +0300)]
scaletempo: add missing G_PARAM_STATIC_STRINGS flags
Canonicalize property names as needed.
Benjamin Otte [Thu, 18 Mar 2010 16:30:26 +0000 (17:30 +0100)]
scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple
Thiago Santos [Thu, 5 Nov 2009 16:40:38 +0000 (13:40 -0300)]
scaletempo: properly update new segments
Scaletempo was missing an update of 'stop' in
new segment parameters when pushing it downstream,
which caused files to end earlier when rate < 1.
Fixes #599903
Based on patch by: Bastian Hecht <hechtb@gmail.com>
Maximilian Högner [Sun, 14 Jun 2009 18:00:51 +0000 (20:00 +0200)]
scaletempo: Explicitely cast to signed integers to fix a segfault
Fixes bug #585660.
Michael Smith [Fri, 13 Feb 2009 20:18:48 +0000 (12:18 -0800)]
scaletempo: Do not use void pointer arithmetic.
Stefan Kost [Thu, 30 Oct 2008 12:13:18 +0000 (12:13 +0000)]
scaletempo: Return the result of parent_class->event()
Original commit message from CVS:
* gst/audiofx/gstscaletempo.c:
Return the result of parent_class->event().
Rov Juvano [Sun, 31 Aug 2008 12:20:33 +0000 (12:20 +0000)]
Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r...
Original commit message from CVS:
Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-scaletempo.xml:
* examples/scaletempo/Makefile.am:
* examples/scaletempo/demo-gui.c: (pop_status_bar),
(status_bar_printf), (demo_gui_seek_bar_format), (update_position),
(demo_gui_seek_bar_change), (demo_gui_do_change_rate),
(demo_gui_do_set_rate), (demo_gui_do_rate_entered),
(demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
(demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
(demo_gui_do_play_pause), (demo_gui_do_open_file),
(demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
(demo_gui_do_about_dialog), (demo_gui_do_quit),
(demo_gui_request_set_stride), (demo_gui_request_set_overlap),
(demo_gui_request_set_search), (demo_gui_rate_changed),
(demo_gui_playing_started), (demo_gui_playing_paused),
(demo_gui_playing_ended), (demo_gui_player_errored),
(demo_gui_stride_changed), (demo_gui_overlap_changed),
(demo_gui_search_changed), (demo_gui_set_player_func),
(demo_gui_set_playlist_func), (build_gvalue_array),
(create_action), (demo_gui_show_func), (demo_gui_set_player),
(demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
(demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
(demo_gui_get_type):
* examples/scaletempo/demo-gui.h:
* examples/scaletempo/demo-main.c: (handle_error_message),
(handle_quit), (main):
* examples/scaletempo/demo-player.c: (no_pipeline),
(demo_player_event_listener), (demo_player_state_changed_cb),
(demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
(demo_player_scale_rate_func), (demo_player_set_rate_func),
(_set_state_and_wait), (demo_player_load_uri_func),
(demo_player_play_func), (demo_player_pause_func), (_seek_to),
(demo_player_seek_by_func), (demo_player_seek_to_func),
(demo_player_get_position_func), (demo_player_get_duration_func),
(demo_player_scale_rate), (demo_player_set_rate),
(demo_player_load_uri), (demo_player_play), (demo_player_pause),
(demo_player_seek_by), (demo_player_seek_to),
(demo_player_get_position), (demo_player_get_duration),
(demo_player_get_property), (demo_player_set_property),
(demo_player_init), (demo_player_class_init),
(demo_player_get_type):
* examples/scaletempo/demo-player.h:
* gst/audiofx/Makefile.am:
* gst/audiofx/gstscaletempo.c: (best_overlap_offset_float),
(best_overlap_offset_s16), (output_overlap_float),
(output_overlap_s16), (fill_queue), (reinit_buffers),
(gst_scaletempo_transform), (gst_scaletempo_transform_size),
(gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
(gst_scaletempo_get_property), (gst_scaletempo_set_property),
(gst_scaletempo_base_init), (gst_scaletempo_class_init),
(gst_scaletempo_init):
* gst/audiofx/gstscaletempo.h:
* gst/audiofx/gstscaletempoplugin.c: (plugin_init):
Add scaletempo plugin, which allows to scale the speed of audio without
changing the pitch by handling seeks with a rate!=1.0.
Integrate it into the docs and add the example application for it.
Fixes bug #537700.
Wim Taymans [Thu, 13 Dec 2012 11:36:20 +0000 (12:36 +0100)]
check: add (but disable) more rtp jitterbuffer tests
Tests need to be ported to 1.0 before they can be enabled but added here so they
don't get forgotten.
See https://bugzilla.gnome.org/show_bug.cgi?id=667838
Havard Graff [Fri, 13 Jan 2012 00:11:31 +0000 (01:11 +0100)]
jitterbuffer: bundle together late lost-events
The scenario where you have a gap in a steady flow of packets of
say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer
will idle up until it receives the first buffer after the gap, but will
then go on to produce 499 lost-events, to "cover up" the gap.
Now this is obviously wrong, since the last possible time for the earliest
lost-events to be played out has obviously expired, but the fact that
the jitterbuffer has a "length", represented with its own latency combined
with the total latency downstream, allows for covering up at least some
of this gap.
So in the case of the "length" being 200ms, while having received packet
500, the jitterbuffer should still create a timeout for packet 491, which
will have its time expire at 10,02 seconds, specially since it might
actually arrive in time! But obviously, waiting for packet 100, that had
its time expire at 2 seconds, (remembering that the current time is 10)
is useless...
The patch will create one "big" lost-event for the first 490 packets,
and then go on to create single ones if they can reach their
playout deadline.
See https://bugzilla.gnome.org/show_bug.cgi?id=667838
Wim Taymans [Thu, 13 Dec 2012 08:27:14 +0000 (09:27 +0100)]
rtspsrc: fix TCP reconnect
Ignore other commands when reconnecting, otherwise the loop function would pause
and the reconnection would not happen. Continue looping after doing a reconnect
so that we have a chance to actually read the new data.
Руслан Ижбулатов [Wed, 12 Dec 2012 21:02:34 +0000 (01:02 +0400)]
directsound, waveform: fix compilation errors caused by circular includes
https://bugzilla.gnome.org/show_bug.cgi?id=690124
Sebastian Dröge [Wed, 12 Dec 2012 17:35:04 +0000 (17:35 +0000)]
ext/sys: Fix some compilation errors caused by circular includes
Philippe Normand [Wed, 12 Dec 2012 11:07:34 +0000 (12:07 +0100)]
deinterleave: properly set srcpad channel position
The src pad caps always describe a single audio channel so only the
first position matters if deinterleave is configured to keep channel
positions in its src pads.
Wim Taymans [Wed, 12 Dec 2012 10:09:42 +0000 (11:09 +0100)]
rtspsrc: timeout on udpsrc is in nanoseconds
Wim Taymans [Wed, 12 Dec 2012 10:08:13 +0000 (11:08 +0100)]
udpsrc: improve timeouts
Make it possible to set the timeout after we went to the READY state by using
the timeout when checking the condition. This also makes it possible to set the
timeout with a higher granularity than seconds.
Wim Taymans [Tue, 11 Dec 2012 12:00:46 +0000 (13:00 +0100)]
deinterlace: add support for strides
Implement stride support correctly by taking it from the GstVideoFrame.
Propose a bufferpool upstream when not operating in passthrough.
Aleix Conchillo Flaque [Thu, 27 Sep 2012 19:17:58 +0000 (12:17 -0700)]
rtspsrc: do not change state to PLAYING if currently chaning state
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
happening in the application thread, so we don't change the state to
PLAYING in the gstrtspsrc thread unless it is safe.
A specific case is when chaning the state to NULL from the application
thread. This will synchronously try to stop the task (with the element
state lock acquired), but we will try a gst_element_set_state from
gstrtspsrc thread which will block on the element state lock causing a
deadlock.
https://bugzilla.gnome.org/show_bug.cgi?id=684312
Alexey Chernov [Mon, 10 Dec 2012 11:44:26 +0000 (11:44 +0000)]
osxvideosink: Fix resizing the Cocoa window on receiving new caps
Fixes bug #689732.
Tim-Philipp Müller [Fri, 30 Nov 2012 20:37:47 +0000 (20:37 +0000)]
v4l2src: link against -lrt for clock_gettime()
Need to explicitly link against -lrt for clock_gettime(), which
we don't get in the libs any more, because core moved the
gmodule-no-export-2.0 bit into Requires.Private.
Not required for newer glibc, but for older ones, so check for that.
Tim-Philipp Müller [Fri, 30 Nov 2012 17:22:59 +0000 (17:22 +0000)]
shout2send: accept audio/webm as well as video/webm
https://bugzilla.gnome.org/show_bug.cgi?id=689336
Tim-Philipp Müller [Fri, 30 Nov 2012 17:20:18 +0000 (17:20 +0000)]
webmux: fix linking with shout2send element
Shout2send only accepts webm format, not matroska, but due
to a bug in matroskamux, webmmux's source pad is also created
with the matroska source pad template as pad template, which
makes the link function think it can't link webmmux to shout2send.
Also add unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=689336
Wim Taymans [Tue, 27 Nov 2012 10:13:37 +0000 (11:13 +0100)]
rtspsrc: use new option parser function
Tim-Philipp Müller [Mon, 26 Nov 2012 15:17:13 +0000 (15:17 +0000)]
law: fix accidental file permissions change
https://bugzilla.gnome.org/show_bug.cgi?id=687469
Tim-Philipp Müller [Sun, 25 Nov 2012 16:05:11 +0000 (16:05 +0000)]
v4l2: remove unused define
Tim-Philipp Müller [Sun, 25 Nov 2012 14:16:09 +0000 (14:16 +0000)]
qtdemux: avoid criticals if unknown fourcc has space at beginning or end
https://bugzilla.gnome.org/show_bug.cgi?id=682936
Tim-Philipp Müller [Sat, 24 Nov 2012 19:32:51 +0000 (19:32 +0000)]
videobox: fix border filling for planar YUV formats
We would get a green border instead of a black one, for
example.
https://bugzilla.gnome.org/show_bug.cgi?id=684991
Tim-Philipp Müller [Sat, 24 Nov 2012 14:27:33 +0000 (14:27 +0000)]
mulaw: const-ify some arrays
Roland Krikava [Fri, 2 Nov 2012 16:38:44 +0000 (12:38 -0400)]
mulawdec: fix integer overrun
There might be more than 65535 samples in a chunk of data.
https://bugzilla.gnome.org/show_bug.cgi?id=687469
Wim Taymans [Thu, 22 Nov 2012 10:34:31 +0000 (11:34 +0100)]
rtspsrc: pause the task instead of spinning
Actually pause the loop task instead of spinning forever.
Joshua M. Doe [Mon, 19 Nov 2012 08:31:37 +0000 (03:31 -0500)]
videoflip: Add gray 8/16 support
Tim-Philipp Müller [Mon, 19 Nov 2012 11:25:14 +0000 (11:25 +0000)]
Automatic update of common submodule
From b497c4f to a72faea
Wim Taymans [Fri, 16 Nov 2012 14:38:29 +0000 (15:38 +0100)]
rtspsrc: handle segment event
Make a segment event when we send a new range header to a client (first PLAY
request or after a seek). Send the segment event in interleaved mode.
Clean the segment event on cleanup
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382
Wim Taymans [Fri, 16 Nov 2012 14:18:07 +0000 (15:18 +0100)]
rtspsrc: fix check for active streams
A stream can be active without a srcpad yet and we want to send
events on those streams as well.
Wim Taymans [Fri, 16 Nov 2012 12:31:04 +0000 (13:31 +0100)]
rtspsrc: create and add pads outside of lock
Create and add the ghostpad for the new stream outside of the lock because it
is not needed and causes deadlocks.
Aleix Conchillo Flaque [Thu, 13 Sep 2012 05:11:20 +0000 (22:11 -0700)]
rtspsrc: allow client to disable reconnection
* gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
rtspsrc always tried to reconnect to the server when the RTSP
connection was closed by the server. This property lets the user
decide whether it wants rtspsrc to reconnect or not.
https://bugzilla.gnome.org/show_bug.cgi?id=683912
Wim Taymans [Fri, 16 Nov 2012 11:16:05 +0000 (12:16 +0100)]
rtspsrc: clear variables before retrying
Else we might unref an old udpsrc twice in cleanup.
Wim Taymans [Fri, 16 Nov 2012 11:00:14 +0000 (12:00 +0100)]
rtspsrc: propose ports in multicast
When the user configured a port-range, propose ports from this range
as the multicast ports. The server is free to ignore this request but if it
honours it, increment our ports so that we suggest the next port pair for the
next stream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420
Wim Taymans [Fri, 16 Nov 2012 10:58:53 +0000 (11:58 +0100)]
rtspsrc: add more debug
Tim-Philipp Müller [Fri, 16 Nov 2012 09:09:38 +0000 (09:09 +0000)]
multifilesink: post messages in max-size mode as well
No reason not to really.
Wim Taymans [Thu, 15 Nov 2012 13:37:44 +0000 (14:37 +0100)]
udpsrc: post error before stopping
Tim-Philipp Müller [Wed, 14 Nov 2012 00:13:36 +0000 (00:13 +0000)]
gst_adapter_prev_timestamp -> gst_adapter_prev_pts
https://bugzilla.gnome.org/show_bug.cgi?id=675598
Nicolas Dufresne [Mon, 12 Nov 2012 18:23:41 +0000 (19:23 +0100)]
videoflip: Add NV12/NV21 support
https://bugzilla.gnome.org/show_bug.cgi?id=688225
Sebastian Dröge [Mon, 12 Nov 2012 12:01:23 +0000 (13:01 +0100)]
vp8enc: Don't leak GstVideoCodecFrames that cause the creation of invisible frames
Fixes bug #682714.
Sebastian Dröge [Mon, 12 Nov 2012 10:47:17 +0000 (11:47 +0100)]
pulse: Use new GType for GThread instead of just G_TYPE_POINTER
Wim Taymans [Mon, 12 Nov 2012 10:14:34 +0000 (11:14 +0100)]
rtpsource: protect against invalid RTP packets
Sebastian Dröge [Mon, 12 Nov 2012 09:44:01 +0000 (10:44 +0100)]
pngdec: Actually use the stop() vfunc implementation
Sebastian Dröge [Mon, 12 Nov 2012 09:31:59 +0000 (10:31 +0100)]
vp8dec: Fix last commit
Sebastian Dröge [Mon, 12 Nov 2012 09:10:15 +0000 (10:10 +0100)]
pngdec: Keep the input state in reset()
It's still valid after a flush and we might not get a new one.
Sebastian Dröge [Mon, 12 Nov 2012 09:08:57 +0000 (10:08 +0100)]
vp8dec: Also destroy decoder in set_format() if it was created already
Fixes a memory leak.
Sebastian Dröge [Mon, 12 Nov 2012 08:48:45 +0000 (09:48 +0100)]
vp8dec: Don't clear input state in reset()
The input state is still valid after flushing until
new caps arrive.
Fixes bug #688092.
Tim-Philipp Müller [Sat, 10 Nov 2012 18:21:28 +0000 (18:21 +0000)]
videocrop: add support for YV12
We can do I420, so we can do YV12 as well.
Alessandro Decina [Sat, 10 Nov 2012 11:39:08 +0000 (12:39 +0100)]
multifilesink: don't write stream headers with key-unit-event
Don't write stream headers, let upstream elements insert them in the stream if
all_headers=true is set in key unit events.
Nicolas Dufresne [Fri, 9 Nov 2012 12:27:16 +0000 (13:27 +0100)]
videocrop: Add NV12/NV21 support
https://bugzilla.gnome.org/show_bug.cgi?id=687964
Debarshi Ray [Fri, 9 Nov 2012 15:31:05 +0000 (16:31 +0100)]
vp8dec: Don't give up so easily if failed to decode a frame
https://bugzilla.gnome.org/show_bug.cgi?id=687436
Sebastian Dröge [Fri, 9 Nov 2012 10:22:30 +0000 (11:22 +0100)]
udpsrc: Also clear GError
Sebastian Dröge [Fri, 9 Nov 2012 10:20:27 +0000 (11:20 +0100)]
udpsrc: Don't error out if we get an ICMP destination-unreachable message when trying to read packets
See bug #529454 and #687782 and commit
751f2bb3646f2beff3698c9f09900dbd0ea08abb
Tim-Philipp Müller [Wed, 7 Nov 2012 20:35:50 +0000 (20:35 +0000)]
configure.ac: update courtesy of autoupdate
Tim-Philipp Müller [Wed, 7 Nov 2012 18:48:49 +0000 (18:48 +0000)]
configure: let AG_GST_PLUGIN_DOCS check for python
And update common for move from AS_PATH_PYTHON to AM_PATH_PYTHON,
which as a side-effect should pick up newer python versions as
well.
https://bugzilla.gnome.org/show_bug.cgi?id=563903
Christian Fredrik Kalager Schaller [Wed, 7 Nov 2012 12:36:33 +0000 (13:36 +0100)]
Fix vp8rtp header names in Makefile
Nicolas Dufresne [Tue, 6 Nov 2012 14:03:55 +0000 (15:03 +0100)]
videocrop: Add support for automatic cropping
This change enable automatic cropping using -1 set to left, top, right or
bottom property. In the case both side are set to automatic cropping, the
croping will be done equally on both side (in the odd case, right and
bottom cropping will be 1 pixel more).
https://bugzilla.gnome.org/show_bug.cgi?id=687761
Debarshi Ray [Fri, 2 Nov 2012 15:39:28 +0000 (16:39 +0100)]
speexdec: Don't unmap or finish_frame an invalid GstBuffer
https://bugzilla.gnome.org/show_bug.cgi?id=687464
Marc Leeman [Tue, 6 Nov 2012 12:22:58 +0000 (13:22 +0100)]
rtsp: the RTCP port number is inclusive
The configured port number pair has its upper bound set to the maximum
allowed RTCP port, inclusive.
See https://bugzilla.gnome.org/show_bug.cgi?id=639420
Tim-Philipp Müller [Sun, 4 Nov 2012 00:07:18 +0000 (00:07 +0000)]
Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
Wim Taymans [Fri, 2 Nov 2012 18:47:26 +0000 (18:47 +0000)]
vrawdepay: don't access rtp buffer after unmap
Read the marker bit before we unmap the rtp packet.
Sebastian Dröge [Fri, 2 Nov 2012 08:34:25 +0000 (09:34 +0100)]
vp8dec: Immediately return if opening the decoder failed
Instead of ignoring any errors.
Debarshi Ray [Thu, 1 Nov 2012 21:02:39 +0000 (22:02 +0100)]
vp8dec: Short circuit gst_vp8_dec_handle_frame if keyframe is missing
https://bugzilla.gnome.org/show_bug.cgi?id=687376
Douglas Bagnall [Thu, 1 Nov 2012 21:53:57 +0000 (10:53 +1300)]
videoconvert: Compare y offset with height, not width, when testing for overlap
This could have prevented images showing that should have when the
source height is greater than its width.
When width exceeds height, as is common, it probably only caused a
miniscule amount of unnecessary work. I haven't tested.
Tim-Philipp Müller [Thu, 1 Nov 2012 21:09:56 +0000 (21:09 +0000)]
rtpvp8: include config.h and minor style fixes
Tim-Philipp Müller [Thu, 1 Nov 2012 20:13:43 +0000 (20:13 +0000)]
rtp: fix tabs/space mess in Makefile.am
Tim-Philipp Müller [Thu, 1 Nov 2012 20:05:49 +0000 (20:05 +0000)]
rtp: move VP8 payloader and depayloader from -bad
Spec is still in draft state, but should hopefully not
change much now. Besides, we announce things as VP8-DRAFT-IETF-01
in our caps, so even if things change in incompatible ways it
should not break anything.
https://bugzilla.gnome.org/show_bug.cgi?id=687263
Tim-Philipp Müller [Wed, 17 Oct 2012 16:34:26 +0000 (17:34 +0100)]
rtpvp8: use gst_element_class_set_static_metadata()
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
Mark Nauwelaerts [Fri, 14 Sep 2012 15:08:49 +0000 (17:08 +0200)]
rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata
Sebastian Dröge [Thu, 5 Apr 2012 16:02:56 +0000 (18:02 +0200)]
rtpvp8: update for GST_PLUGIN_DEFINE() API changes
Wim Taymans [Wed, 28 Mar 2012 10:49:54 +0000 (12:49 +0200)]
rtpvp8: update for buffer changes
Danilo Cesar Lemes de Paula [Thu, 1 Mar 2012 17:59:55 +0000 (14:59 -0300)]
rtpvp8; fix compatibility with the third draft
https://bugzilla.gnome.org/show_bug.cgi?id=671073
Mark Nauwelaerts [Wed, 25 Jan 2012 15:20:41 +0000 (16:20 +0100)]
rtpvp8: port some more to new memory API
Olivier Crête [Wed, 25 Jan 2012 09:45:51 +0000 (10:45 +0100)]
rtpvp8: port to 0.11
Sebastian Dröge [Mon, 3 Oct 2011 10:06:27 +0000 (12:06 +0200)]
rtpvp8pay: Fix typo
Youness Alaoui [Fri, 23 Sep 2011 22:58:30 +0000 (22:58 +0000)]
rtpvp8: Update the pay/depay to the ietf-draft-01 spec
Vincent Penquerc'h [Sat, 10 Sep 2011 10:31:20 +0000 (11:31 +0100)]
rtpvp8: fix bitstream parsing using the wrong kind of bitreader
VP8 uses a probabilistic bool coder, not a straight bit coder.
This fixes parsing when error-resilient is set.
This commit includes a copy of libvpx's bool coder, BSD licensed.
https://bugzilla.gnome.org/show_bug.cgi?id=652694
Olivier Crête [Tue, 12 Jul 2011 22:03:53 +0000 (18:03 -0400)]
rtpvp8: Reject unknown bitstream versions
Edward Hervey [Fri, 4 Mar 2011 10:59:44 +0000 (11:59 +0100)]
rtpvp8: Fix unitialized variable
Makes macosx compiler happy.
Sjoerd Simons [Sun, 23 Jan 2011 17:02:38 +0000 (17:02 +0000)]
rtpvp8depay: Accept packets with only one byte of data
When fragmenting partions it can happen that an RTP packet only caries 1
byte of RTP data.
Sjoerd Simons [Sun, 23 Jan 2011 16:42:17 +0000 (16:42 +0000)]
rtpvp8pay: Treat the frame header just like any other partition
When setting up the initial mapping just act as if the global frame
information is another partition. This saves special-casing it later in
the actual packetizing code.
Sjoerd Simons [Sun, 16 May 2010 16:23:17 +0000 (17:23 +0100)]
rtpvp8: Add simple payloaders and depayloaders for VP8
Minimal implementation of http://www.webmproject.org/code/specs/rtp/,
version 0.3.2
Wim Taymans [Thu, 1 Nov 2012 18:42:39 +0000 (18:42 +0000)]
gstpay: fix for 1.0 events
Caps events are sometimes not followed by a buffer but by an event. Flush any
pending caps before we make a packet with the event.
Chain up to the parent event handler before we attempt to push RTP packets, it
might be a segment event.
Wim Taymans [Thu, 1 Nov 2012 18:42:24 +0000 (18:42 +0000)]
gstdepay: fix small leak
Wim Taymans [Thu, 1 Nov 2012 17:44:11 +0000 (17:44 +0000)]
gstdepay: add support for events
Conflicts:
gst/rtp/gstrtpgstdepay.c
Wim Taymans [Thu, 1 Nov 2012 17:40:31 +0000 (17:40 +0000)]
rtpgstpay: add support for sending events
We currently only send tags and custom events. The other events
might interfere with the receiver timings or are otherwise handled
by RTP.
Conflicts:
gst/rtp/gstrtpgstpay.c
Wim Taymans [Thu, 1 Nov 2012 15:54:58 +0000 (15:54 +0000)]
gstpay: rewrite payloader
Use adapter to assemble the payload and make a flush function to
turn this payload into (fragmented) packets.
Conflicts:
gst/rtp/gstrtpgstpay.c
gst/rtp/gstrtpgstpay.h
Douglas Bagnall [Thu, 1 Nov 2012 13:03:44 +0000 (13:03 +0000)]
videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH
https://bugzilla.gnome.org/show_bug.cgi?id=687330
Douglas Bagnall [Thu, 1 Nov 2012 13:02:16 +0000 (13:02 +0000)]
videbox: fix border filling for gray formats
Get the height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH.
https://bugzilla.gnome.org/show_bug.cgi?id=687330
Wim Taymans [Thu, 1 Nov 2012 11:58:57 +0000 (11:58 +0000)]
gstdepay: check for correct fragment offset
Make sure we only insert the rtp packet in the adapter when the
frag_offset matches. When the first packet of a fragment is dropped,
it avoids putting the remaining packets in the adapter and processing
the partial fragment.
Conflicts:
gst/rtp/gstrtpgstdepay.c
Wim Taymans [Thu, 1 Nov 2012 11:54:50 +0000 (11:54 +0000)]
gstpay: set C flag on all buffers of the fragment
Set the C flags on all the fragments instead of only those with
caps in them. This makes it easier in the receiver to check if there
is a caps in the assembled fragments just by looking at the last RTP
packet flags.
Wim Taymans [Thu, 1 Nov 2012 10:55:03 +0000 (10:55 +0000)]
gstdepay: use the capsversion
Take the caps from the input caps and store it in the slot given
by capsversion.