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Author SHA1 Message Date
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