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6440 commits

Author SHA1 Message Date
Youness Alaoui
f0e209b638 rtpmux: missing comment 
20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz
 2012-12-16 16:30:33 +00:00
Olivier Crete
3ed5590da6 rtpmux: Make buffer writable before writing into it 
20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz
 2012-12-16 16:30:31 +00:00
Olivier Crete
dd13f7c8ef rtpmux: Set pads active when adding them to a potentially running element 
20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz
 2012-12-16 16:30:27 +00:00
Olivier Crete
1c5075f927 rtpmux: Fix multiple ref leaks (patches by SP GLE) 
20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz
 2012-12-16 16:30:23 +00:00
Zeeshan Ali
42f455e902 rtpmux: send event to all src pads 
20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz
 2012-12-16 16:30:18 +00:00
Zeeshan Ali
dba101bb0f rtpmux: print a warning if receive an error iterating sinkpads 
20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz
 2012-12-16 16:30:15 +00:00
Zeeshan Ali
baa48dc6bc rtpmux: deal with all the gst_iterator_next() return values 
20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz
 2012-12-16 16:30:12 +00:00
Zeeshan Ali
de40874670 rtpmux: Return correct value from the event handler 
20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz
 2012-12-16 16:30:08 +00:00
Zeeshan Ali
ed76f67e96 rtpmux: Ville's original patch to fix the traversal of dtmf event 
20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz
 2012-12-16 16:30:05 +00:00
zeeshan.ali@nokia.com
94ebe07862 rtpmux: Set the correct ts-offset on the get_prop value 
20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz
 2012-12-16 16:30:01 +00:00
zeeshan.ali@nokia.com
1ee542c378 rtpmux: Refactorize state_change 
20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz
 2012-12-16 16:29:58 +00:00
zeeshan.ali@nokia.com
2498ba671a rtpmux: set SSRC on the packets 
20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz
 2012-12-16 16:29:55 +00:00
zeeshan.ali@nokia.com
ee69c2690d rtpmux: Code clean-up and more debug output 
20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz
 2012-12-16 16:29:52 +00:00
zeeshan.ali@nokia.com
1c799ce964 rtpmux: Use own clock-base 
20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz
 2012-12-16 16:29:48 +00:00
zeeshan.ali@nokia.com
b04630d7a2 rtpmux: Only accept RTP streams that have the same clock-rate 
20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz
 2012-12-16 16:29:45 +00:00
zeeshan.ali@nokia.com
6fe1e02efd rtpmux: Some more code-cleanups 
20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz
 2012-12-16 16:29:42 +00:00
zeeshan.ali@nokia.com
1603223ee5 rtpmux: return newpad instead of NULL and warn if failed to create a pad 
20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz
 2012-12-16 16:29:38 +00:00
zeeshan.ali@nokia.com
23d3ed5c5f rtpmux: Refactorize the RTPMux code 
20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz
 2012-12-16 16:29:35 +00:00
zeeshan.ali@nokia.com
21e6e951f6 rtpmux: Some more doc fixing 
20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz
 2012-12-16 16:29:32 +00:00
zeeshan.ali@nokia.com
0de7fb6f37 rtpmux: More Refactoring 
20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz
 2012-12-16 16:29:29 +00:00
zeeshan.ali@nokia.com
0f755657ce rtpmux: More documentation 
20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz
 2012-12-16 16:29:26 +00:00
zeeshan.ali@nokia.com
5483c78ac0 rtpmux: Refactor the event handler function 
20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz
 2012-12-16 16:29:23 +00:00
zeeshan.ali@nokia.com
db1523ae60 rtpmux: Add RTPDTMFMux element 
20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz
 2012-12-16 16:29:19 +00:00
zeeshan.ali@nokia.com
97ff54dce7 rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable 
20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz
 2012-12-16 16:29:16 +00:00
zeeshan.ali@nokia.com
1a227ac7e5 rtpmux: Put more helpful description 
20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz
 2012-12-16 16:29:13 +00:00
zeeshan.ali@nokia.com
d876c0d8cc rtpmux: remove the (commented-out) code for blocking the pads 
20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz
 2012-12-16 16:29:10 +00:00
zeeshan.ali@nokia.com
209228c44d rtpmux: Drop buffers instead of blocking the sinkpads 
20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz
 2012-12-16 16:29:05 +00:00
zeeshan.ali@nokia.com
795822ffa5 rtpmux: Implement stream locking, needed for DTMF 
20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz
 2012-12-16 16:29:02 +00:00
zeeshan.ali@nokia.com
fd209faa56 rtpmux: use GST_*_OBJECT instead of g_* 
20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz
 2012-12-16 16:28:58 +00:00
zeeshan.ali@nokia.com
b0208cb0a6 rtpmux: No need to manage pads, parent does that for us 
20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz
 2012-12-16 16:28:55 +00:00
zeenix@gmail.com
74e9071dad rtpmux: Fix copyright header 
20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz
 2012-12-16 16:28:51 +00:00
zeeshan.ali@nokia.com
3c4cdf1541 rtpmux: The first implementation of RTP muxer 
20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz
 2012-12-16 16:28:41 +00:00
Tim-Philipp Müller
b19122bac8 scaletempo: no need for a private struct 2012-12-15 21:27:01 +00:00
Tim-Philipp Müller
61913ab7b4 audiofx: move scaletempo element from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=687262
 2012-12-14 13:16:17 +00:00
Sebastian Dröge
314765c294 scaletempo: Fix event leak 2012-12-14 13:16:17 +00:00
Sebastian Dröge
490e408991 scaletempo: Fix timestamp tracking 2012-12-14 13:16:17 +00:00
Sebastian Dröge
502eb8d1b7 scaletempo: Implement LATENCY query 2012-12-14 13:16:17 +00:00
Sebastian Dröge
c7589817cb scaletempo: Store instance private data in the instance struct 
Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE()
is really slow.
 2012-12-14 13:16:17 +00:00
Tim-Philipp Müller
e552bd484f scaletempo: 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.
 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
d2dd91ac47 scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-12-14 13:16:17 +00:00
Wim Taymans
cb1743d578 scaletempo: ffmpegcolorspace is no more 2012-12-14 13:16:17 +00:00
Sebastian Dröge
93e1091d7f scaletempo: Update for GST_PLUGIN_DEFINE() API changes 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
3286cdd542 scaletempo: port to 0.11 2012-12-14 13:16:16 +00:00
Stefan Kost
62d780cd51 scaletempo: improve the docs 
Fix the syntax, add more explanation and xref the properties.
 2012-12-14 13:16:16 +00:00
Chris E Jones
caf2b6cb5c scaletempo: Correctly handle newsegment events with stop==-1 
Fixes bug #645420.
 2012-12-14 13:16:16 +00:00
Stefan Kost
6d54058982 scaletempo: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-14 13:16:16 +00:00
Benjamin Otte
38bc2dfb4a scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-14 13:16:16 +00:00
Thiago Santos
2d72ec153a 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>
 2012-12-14 13:16:16 +00:00
Maximilian Högner
2fe7a97f1c scaletempo: Explicitely cast to signed integers to fix a segfault 
Fixes bug #585660.
 2012-12-14 13:16:16 +00:00
Michael Smith
1b1f6f56d6 scaletempo: Do not use void pointer arithmetic. 2012-12-14 13:16:16 +00:00
Stefan Kost
9284c85b33 scaletempo: Return the result of parent_class->event() 
Original commit message from CVS:
* gst/audiofx/gstscaletempo.c:
Return the result of parent_class->event().
 2012-12-14 13:16:16 +00:00
Rov Juvano
43e79f7769 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.
 2012-12-14 13:16:15 +00:00
Havard Graff
9c94f1187c 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
 2012-12-13 12:00:43 +01:00
Wim Taymans
a858bf46db 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.
 2012-12-13 09:30:59 +01:00
Philippe Normand
a8fa9f2b47 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.
 2012-12-12 11:20:56 +00:00
Wim Taymans
b1dc816772 rtspsrc: timeout on udpsrc is in nanoseconds 2012-12-12 11:09:42 +01:00
Wim Taymans
32bd981303 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.
 2012-12-12 11:08:13 +01:00
Wim Taymans
abd7e33db6 deinterlace: add support for strides 
Implement stride support correctly by taking it from the GstVideoFrame.
Propose a bufferpool upstream when not operating in passthrough.
 2012-12-11 13:00:46 +01:00
Aleix Conchillo Flaque
3503aef946 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
 2012-12-10 15:13:22 +01:00
Tim-Philipp Müller
672ab8fb5b 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
 2012-11-30 17:22:34 +00:00
Wim Taymans
64cdbb77a9 rtspsrc: use new option parser function 2012-11-27 11:13:37 +01:00
Tim-Philipp Müller
5dee61a8d5 law: fix accidental file permissions change 
https://bugzilla.gnome.org/show_bug.cgi?id=687469
 2012-11-26 15:17:13 +00:00
Tim-Philipp Müller
314efb684b qtdemux: avoid criticals if unknown fourcc has space at beginning or end 
https://bugzilla.gnome.org/show_bug.cgi?id=682936
 2012-11-25 14:16:09 +00:00
Tim-Philipp Müller
efaa80fbc6 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
 2012-11-24 19:32:51 +00:00
Tim-Philipp Müller
ef6c16a32e mulaw: const-ify some arrays 2012-11-24 14:27:33 +00:00
Roland Krikava
3be45f7022 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
 2012-11-24 14:24:41 +00:00
Wim Taymans
5d0507c09e rtspsrc: pause the task instead of spinning 
Actually pause the loop task instead of spinning forever.
 2012-11-22 11:34:31 +01:00
Joshua M. Doe
fe9fb8d8a7 videoflip: Add gray 8/16 support 2012-11-20 12:49:49 +01:00
Wim Taymans
c28bfa8902 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
 2012-11-16 15:38:29 +01:00
Wim Taymans
bd91bd3193 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.
 2012-11-16 15:22:46 +01:00
Wim Taymans
11cf4d4fd3 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.
 2012-11-16 13:33:44 +01:00
Aleix Conchillo Flaque
6c855edf03 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
 2012-11-16 12:55:10 +01:00
Wim Taymans
e2a4d28c1f rtspsrc: clear variables before retrying 
Else we might unref an old udpsrc twice in cleanup.
 2012-11-16 12:17:37 +01:00
Wim Taymans
cc9cb26be1 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
 2012-11-16 12:17:37 +01:00
Wim Taymans
5025b3f1b3 rtspsrc: add more debug 2012-11-16 12:17:37 +01:00
Tim-Philipp Müller
6f1aa3e4d5 multifilesink: post messages in max-size mode as well 
No reason not to really.
 2012-11-16 09:13:22 +00:00
Wim Taymans
c33507f186 udpsrc: post error before stopping 2012-11-15 14:48:59 +01:00
Tim-Philipp Müller
bdf3c77828 gst_adapter_prev_timestamp -> gst_adapter_prev_pts 
https://bugzilla.gnome.org/show_bug.cgi?id=675598
 2012-11-14 00:13:36 +00:00
Nicolas Dufresne
673d2d24b8 videoflip: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=688225
 2012-11-13 14:25:04 +01:00
Wim Taymans
c755af0cb0 rtpsource: protect against invalid RTP packets 2012-11-12 11:18:30 +01:00
Tim-Philipp Müller
35fafae241 videocrop: add support for YV12 
We can do I420, so we can do YV12 as well.
 2012-11-10 18:21:28 +00:00
Alessandro Decina
b916d2b398 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.
 2012-11-10 12:41:33 +01:00
Nicolas Dufresne
e111068f7b videocrop: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=687964
 2012-11-10 01:52:44 +01:00
Sebastian Dröge
c70ba7765a udpsrc: Also clear GError 2012-11-09 11:22:30 +01:00
Sebastian Dröge
b86d20e45b 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
751f2bb364
 2012-11-09 11:20:27 +01:00
Christian Fredrik Kalager Schaller
485505f323 Fix vp8rtp header names in Makefile 2012-11-07 13:36:33 +01:00
Nicolas Dufresne
1ad8ebac44 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
 2012-11-07 11:20:24 +01:00
Marc Leeman
7cbca3dcd1 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
 2012-11-06 13:22:58 +01:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Wim Taymans
9857e6af4d vrawdepay: don't access rtp buffer after unmap 
Read the marker bit before we unmap the rtp packet.
 2012-11-02 18:48:17 +00:00
Douglas Bagnall
0b898ab911 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.
 2012-11-02 09:29:30 +01:00
Tim-Philipp Müller
5ac789408b rtpvp8: include config.h and minor style fixes 2012-11-01 21:10:21 +00:00
Tim-Philipp Müller
4a849d6690 rtp: fix tabs/space mess in Makefile.am 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
321acd14dc 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
 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
44efab8e3d 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.
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
bc7dbbbd4f rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-11-01 20:53:48 +00:00
Sebastian Dröge
4853001547 rtpvp8: update for GST_PLUGIN_DEFINE() API changes 2012-11-01 20:53:48 +00:00
Wim Taymans
fccfca38d4 rtpvp8: update for buffer changes 2012-11-01 20:53:48 +00:00
Danilo Cesar Lemes de Paula
3edffb13e3 rtpvp8; fix compatibility with the third draft 
https://bugzilla.gnome.org/show_bug.cgi?id=671073
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
d9581832a0 rtpvp8: port some more to new memory API 2012-11-01 20:53:47 +00:00
Olivier Crête
c6761daa27 rtpvp8: port to 0.11 2012-11-01 20:53:47 +00:00
Sebastian Dröge
2c5ea76bdc rtpvp8pay: Fix typo 2012-11-01 20:53:47 +00:00
Youness Alaoui
1cf155d70d rtpvp8: Update the pay/depay to the ietf-draft-01 spec 2012-11-01 20:53:47 +00:00
Vincent Penquerc'h
88aade4150 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
 2012-11-01 20:53:47 +00:00
Olivier Crête
97c3f3617c rtpvp8: Reject unknown bitstream versions 2012-11-01 20:53:47 +00:00
Edward Hervey
74a1a704bf rtpvp8: Fix unitialized variable 
Makes macosx compiler happy.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
6ed6318076 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.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
a45e7a3fc0 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.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
e9f4e9342f rtpvp8: Add simple payloaders and depayloaders for VP8 
Minimal implementation of http://www.webmproject.org/code/specs/rtp/,
version 0.3.2
 2012-11-01 20:53:47 +00:00
Wim Taymans
d6fd0ebd04 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.
 2012-11-01 18:42:39 +00:00
Wim Taymans
05232c55a5 gstdepay: fix small leak 2012-11-01 18:42:24 +00:00
Wim Taymans
08e5a197b4 gstdepay: add support for events 
Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 18:18:19 +00:00
Wim Taymans
54b783b5a3 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
 2012-11-01 18:06:11 +00:00
Wim Taymans
6502d08e43 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
 2012-11-01 17:57:52 +00:00
Douglas Bagnall
e3c77ba709 videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH 
https://bugzilla.gnome.org/show_bug.cgi?id=687330
 2012-11-01 13:03:44 +00:00
Douglas Bagnall
79403bcb0c 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
 2012-11-01 13:02:16 +00:00
Wim Taymans
c0713e4b80 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
 2012-11-01 12:09:47 +00:00
Wim Taymans
8a402e0c06 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.
 2012-11-01 12:06:08 +00:00
Wim Taymans
d78ff07f7d gstdepay: use the capsversion 
Take the caps from the input caps and store it in the slot given
by capsversion.
 2012-11-01 11:37:44 +00:00
Wim Taymans
936c3819b5 gstpay: send caps inline 
Place the capsversion on the outgoing caps so that they end up in
an SDP as well. Receivers need to know what capsversion a particular
caps is for to be able to match the caps to the CV in the RTP packets.
Place the caps inside the RTP packet whenever the caps change.

Based on patch by Andrzej Bieniek <andrzej.bieniek@pure.com>

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
 2012-11-01 11:34:33 +00:00
Andrzej Bieniek
3b1931a039 gstpay: add debug 
Conflicts:
	gst/rtp/gstrtpgstpay.c
 2012-11-01 11:28:50 +00:00
Andrzej Bieniek
ee5ecc7773 depay: correctly skip caps header size 
Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 11:27:13 +00:00
Tim-Philipp Müller
ef0805ea14 matroskademux: put streamheaders on vorbis/speex/flac/theora caps to make remuxing work 
https://bugzilla.gnome.org/show_bug.cgi?id=640589
 2012-10-30 23:29:46 +00:00
Tim-Philipp Müller
752cf98745 gst: fix variable order in some Makefile.am 
https://bugzilla.gnome.org/show_bug.cgi?id=687013
 2012-10-27 23:27:38 +01:00
Antoine Tremblay
a1c86de09a gst: add various missing GST_PLUGINS_BASE_LIBS in Makefile.am 
Those plugins depend on either libgstaudio or libgstvideo,
which are in gst-plugins-base.

https://bugzilla.gnome.org/show_bug.cgi?id=687013
 2012-10-27 23:26:41 +01:00
Alexey Fisher
29cd24bc41 matroskademux: mark invisible VP8 frames with the DECODE_ONLY flag 
https://bugzilla.gnome.org/show_bug.cgi?id=654259
 2012-10-27 14:46:02 +01:00
Stas Sergeev
238a5ec826 multifilesrc: fix stop index handling 
Make sure the stop index is always honoured. Avoids
endless loop if one wants to read and output the same
file N times, for example.

https://bugzilla.gnome.org/show_bug.cgi?id=654853
 2012-10-26 11:04:01 +01:00
Руслан Ижбулатов
78193dfe71 matroskademux: Support recursive SimpleTags 
Fixes #682644
Depends on #682615
 2012-10-26 10:16:42 +02:00
Руслан Ижбулатов
cd719bb808 matroskademux: Expand the tag mapping. 
* Also expose unknown tags as key=value pairs.
* Arrange tag map in the same order tags are listed in Matroska spec, leaving
unmapped tags as comments.
* More specific TODOs.
* Remove duplicate DATE define.

Fixes #682615
Depends on #682524
 2012-10-26 10:12:52 +02:00
Sebastian Dröge
6c635ce64f matroskademux: Fix uninitialized variable compiler warning 2012-10-26 10:09:39 +02:00
Руслан Ижбулатов
71fd688ef0 matroskademux: Matroska tag TargetType support 
* Reads TargetType and TargetTypeValue from a Tag.
* After Tag is completely read, processes taglist, substituting some of the
tags depending on target type value and the presence of video/subtitle streams.
* Supports reading two new simpletags - PART_NUMBER and TOTAL_PARTS

Depends on #682448
Fixes #682524
 2012-10-26 10:08:18 +02:00
Руслан Ижбулатов
b75628f041 matroskademux: Per-track tags for Matroska 
Requires Matroska file to have sane layout (track info before tag info).
Uses replace-merge.
Makes track UIDs 64-bit.

Fixes #682448
 2012-10-26 10:03:55 +02:00
Tim-Philipp Müller
fe7236230c multifilesrc: fix typo in property description 2012-10-25 20:19:44 +01:00
Michael Smith
a88caf84b4 qtdemux: read video format header fully (so we can find 'pasp' atoms) for more fourccs. 
Fixes aspect ratio of prores files.
 2012-10-25 12:18:50 -07:00
Thiago Santos
02d91dcd24 imagefreeze: the new get_caps already does the filter intersection 
It should be faster to pass the caps to intersect as the filter caps,
rather than using NULL and intersecting 'manually' later.

https://bugzilla.gnome.org/show_bug.cgi?id=686837
 2012-10-25 10:32:17 -03:00
Thiago Santos
115581eb2e imagefreeze: avoid assertion when using accept caps query 
This query must receive a fixed caps, so imagefreeze should
fixate its framerate before sending the query downstream.

https://bugzilla.gnome.org/show_bug.cgi?id=686837
 2012-10-25 09:39:36 -03:00
Arnaud Vrac
bc79fe565c qtdemux: use correct type for channel-mask bitmask 
Fixes crash on 32-bit systems.
 2012-10-24 12:54:08 +01:00
Tim-Philipp Müller
7275860bdd flacparse: fix coverart extraction if vorbis comments come after picture header 
See sample file for bug #684701.
 2012-10-23 16:02:05 +01:00
Tim-Philipp Müller
7c41f42eec flacparse: ignore bad headers if we have a valid STREAMINFO header 
If we run into any header parsing issues and we have a valid
STREAMINFO header already, don't error out, but just stop
header parsing and try to find some audio frames.

https://bugzilla.gnome.org/show_bug.cgi?id=684701
 2012-10-23 13:56:54 +01:00
Tim-Philipp Müller
49cc719809 flacparse: post proper error message and fix buffer leak on header parsing error 
https://bugzilla.gnome.org/show_bug.cgi?id=684701
 2012-10-23 13:56:54 +01:00
Michael Smith
150bd97e96 qtdemux: with raw audio, set a default channel-mask for multichannel audio. 
This doesn't actually parse 'chan' because it's absurdly complex.
 2012-10-22 22:34:43 -07:00
Sebastian Rasmussen
9fc62a58e3 updsrc: fix typo causing compilation error 
gstudpsrc.c: In function 'gst_udpsrc_create':
gstudpsrc.c:365: error: 'ret' may be used uninitialized in this function

https://bugzilla.gnome.org/show_bug.cgi?id=686642
 2012-10-22 23:19:28 +01:00
Wim Taymans
a2eead3d60 avi_ fix invert function 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686550
 2012-10-22 11:55:59 +02:00
Wim Taymans
0e3ef30c31 avi: fix debug 2012-10-22 11:55:22 +02:00
Wim Taymans
199aaa4021 qtdemux: add support for 'generic' samples 
Add support for stuffing a complete stream into 1 sample.

See https://bugzilla.gnome.org/show_bug.cgi?id=686550
 2012-10-22 11:39:37 +02:00
Tim-Philipp Müller
aa3ba65eb5 qtdemux: don't leak gst_riff_strf_auds in case of MS/RIFF audio 
https://bugzilla.gnome.org/show_bug.cgi?id=681192
 2012-10-19 19:26:45 +01:00
Mark Nauwelaerts
35cd53867c matroskamux: unsigned subtitle template 2012-10-19 16:14:01 +02:00
Youness Alaoui
13328bc129 videomixer2: Fix race condition where a src setcaps is ignored 
If both pads receive data at the same time, they will both get their
sink_setcaps called which will call the src_setcaps, but there is
a race condition where the second one might not be called.
Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=683842
 2012-10-19 12:10:31 +02:00
Mark Nauwelaerts
5742352e10 matroskamux: do not use unoffical V_MJPEG codec id 
Since it's not spec'ed, consider it a VfW compatibility
case. Many applications (e.g. avidemux) don't understand
the unofficial V_MJPEG id.

Fixes #659837.

Conflicts:
	gst/matroska/matroska-mux.c
 2012-10-18 18:29:40 +01:00
Wim Taymans
e9040e90a5 jpegdepay: store quant tables in zigzag order 2012-10-17 14:23:01 +02:00