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52 commits

Author SHA1 Message Date
François Laignel
39f0905a7e Use gst_element_request_pad_simple 
Instead of the deprecated gst_element_get_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/958>
 2021-05-05 06:17:20 +00:00
Tim-Philipp Müller
c9a47c0c8d Remove autotools build system 2019-10-14 11:04:18 +01:00
Aaron Boxer
46989dca96 documentation: fix a number of typos 2019-10-05 22:38:11 +00:00
Tim-Philipp Müller
6b68b73341 tests: .gitignore more test and example binaries 2019-03-06 17:26:03 +00:00
Mathieu Duponchelle
f52e16ceb8 Revert "rtpbin: receive bundle support" 
This reverts commit dcd3ce9751.

This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.

This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.

Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.

Fixes #537
 2018-12-20 13:25:10 +00:00
Tim-Philipp Müller
9f678898f8 meson: build examples 
https://bugzilla.gnome.org/show_bug.cgi?id=784134
 2017-06-26 11:10:29 +01:00
Sebastian Dröge
9f5fe2673e rtp: Remove unused variable in example 
client-PCMA.c:84:22: warning: unused variable 'isrc' [-Wunused-variable]
  GObject *session, *isrc, *osrc;
                     ^
 2017-01-25 20:56:24 +02:00
Philippe Normand
dcd3ce9751 rtpbin: receive bundle support 
A new signal named on-bundled-ssrc is provided and can be
used by the application to redirect a stream to a different
GstRtpSession or to keep the RTX stream grouped within the
GstRtpSession of the same media type.

https://bugzilla.gnome.org/show_bug.cgi?id=772740
 2016-11-16 08:56:34 +01:00
Gaurav Gupta
6542edd909 tests: Fix memory leak in test rtpaux test 
https://bugzilla.gnome.org/show_bug.cgi?id=772496
 2016-10-06 13:23:28 +03:00
Olivier Crête
a390a6791c rtp example: Fix leak 
Also stop fetching the internal source as this
functionality has been broken.
 2016-07-11 11:59:21 -04:00
Sebastian Dröge
b3dae8c969 rtp: Add examples with VTS/ATS for VP8/OPUS 
Let's have an example with modern codecs.
 2015-07-01 12:42:40 +02:00
Sebastian Dröge
7bd1cfa197 examples: Set RTP profile to AVPF for rtpaux examples 
https://bugzilla.gnome.org/show_bug.cgi?id=746543
 2015-06-02 11:38:15 +02:00
Tim-Philipp Müller
2e412a447a docs: update example pipelines in element docs 
Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
 2015-05-10 11:05:00 +01:00
Henning Heinold
8aa2630068 examples: port python rtp PCMA client/server tests to 1.0 
https://bugzilla.gnome.org/show_bug.cgi?id=739930
 2014-11-17 00:23:13 +00:00
Sebastian Rasmussen
5cd0261f77 examples: client-rtpaux: Release reference to parent when done 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732976
 2014-07-10 22:25:25 +01:00
George Kiagiadakis
a7823bc522 examples/*-rtpaux: specify payload type association for the audio stream, so that rtx works also for audio 2014-01-15 10:13:12 +01:00
George Kiagiadakis
9226091235 rtprtxreceive: modify to use a payload-type map like rtprtxsend 2014-01-03 20:48:29 +01:00
Wim Taymans
130ad1b1fa rtprtxsend: Allow SSRC-multiplexing and multiple payload types in the original stream 
Conflicts:
	tests/examples/rtp/server-rtpaux.c
 2014-01-03 20:48:29 +01:00
Torrie Fischer
e29b5f8b41 examples: rtp: Add end-to-end rtpbin example with RTX elements 
This example demonstrates how to use rtpbin with retransmission (rtx)
elements set in the place of rtpbin's "aux" elements in order to
enable RTP retransmission according to the rules of RFC4588.
 2014-01-03 20:48:29 +01:00
Wim Taymans
aa9af2c82e examples: we don't need the queue anymore 2013-09-16 15:55:55 +02:00
Wim Taymans
43359b9244 tests: add retransmission example 2013-08-23 12:10:19 +02:00
B.Prathibha
7bb368ee4c tests: use g_timeout_add_seconds instead of g_timeout_add 
https://bugzilla.gnome.org/show_bug.cgi?id=692615
 2013-01-27 15:38:12 +00:00
Tim-Philipp Müller
1a073355bc examples: check for uri argument in decodebin-h264p-amr server example 
Otherwise people get a rather confusing error message.
 2012-12-31 18:59:18 +00:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Tim-Philipp Müller
3bbbcd266c examples: update some element names for 1.0 in RTP examples 
gstrtpbin -> rtpbin
ffdec_*   -> avdec_*
ffenc_*   -> avenc_*
 2012-10-11 22:36:21 +01:00
Wim Taymans
829c80ce6c fix more caps 2012-09-14 13:30:37 +02:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Mark Nauwelaerts
f189f62b13 Merge branch 'master' into 0.11 
Conflicts:
	ext/wavpack/gstwavpackenc.c
	tests/check/elements/audioiirfilter.c
	tests/examples/v4l2/probe.c
 2012-03-01 11:29:50 +01:00
Edward Hervey
9beda57c3a Suppress deprecation warnings in selected files, for g_value_array_* mostly 2012-02-27 14:47:25 +01:00
Tim-Philipp Müller
0f3b7b010e build: ignore GValueArray deprecation warnings for the time being 
until this gets sorted out with the GLib folks and we have a
viable alternative.

https://bugzilla.gnome.org/show_bug.cgi?id=667228
 2012-02-01 16:40:51 +00:00
Tim-Philipp Müller
09ca5fa910 rtpmanager: rename gstrtp* -> rtp* 
This was done in 0.10 to avoid conflict with the rtp elements in
farsight, but the gst-prefixing is no longer needed in 0.11
 2011-11-24 00:54:08 +00:00
Tim-Philipp Müller
b005dfea5b examples: fix some warnings in rtp example 
Caused by -DG_DISABLE_ASSERT
 2011-04-16 18:07:35 +01:00
Stefan Kost
4e9daf0e49 example: fix the variable name for the ip-address 
Fix the name in the launch pipeline and use a value of "localhost" by default.
 2011-03-01 22:40:19 +02:00
Stefan Kost
f73cc87239 rtp-examples: move capsfilter behind converters 
We need to have the capsfilter behin the converters to make the converters
convert from the formats v4l2src can do to what we request with the
capsfilter.
 2011-02-22 14:54:28 +02:00
Stefan Kost
768dbfaf92 rtp-examples: fix ascii-art 
Some boxes where misaligned due to long "audiotetssrc" name. Trim trailing
whitespace.
 2011-02-22 14:54:28 +02:00
Tim-Philipp Müller
285235a10a examples: autoaudisink -> autoaudiosink in RTP examples 2011-01-24 13:39:58 +00:00
Wim Taymans
17c45a8869 examples: add example RTP stats 
Add some more RTP examples for how to retrieve RTP stats in a receiver.
 2010-12-23 13:58:30 +01:00
Wim Taymans
fdfe76ac53 examples: improve RTP examples 
Make the examples use autovideosink and ffmpegcolorspace for better
compàtibility.
Make some more variables for the sink and the decoders.
Set zerolatency tuning on x264enc for better realtime results.
 2010-12-02 19:16:47 +01:00
Wim Taymans
1e310bc1ee test: add python version of the audio sender 
Add a python version of the audio sender pipeline.

Ported by Sp4rc on IRC.
 2010-10-04 17:56:57 +02:00
Wim Taymans
b50ce27b14 tests: Add python RTP client example 
Add a python version of the PCMA client app.

Ported by Sp4rc on IRC.
 2010-10-04 17:52:22 +02:00
Thijs Vermeir
9c429de37a examples: fix indentation on rtp client example 2010-09-30 11:38:38 +02:00
Thijs Vermeir
92a1adfde8 examples: fix typo in port of rtp examples 2010-09-30 11:38:38 +02:00
Michael Smith
e98c682732 rtp examples: remove executable bits from C files. 2009-10-23 17:25:17 -07:00
Sebastian Dröge
1a291a126a rtp: Use autoaudio{sink,src} instead of alsa in the examples 2009-09-30 18:46:57 +02:00
Wim Taymans
89060e8696 tests/examples/rtp/server-alsasrc-PCMA.c: Add some example code for printing the RTP manager stats. 
Original commit message from CVS:
* tests/examples/rtp/server-alsasrc-PCMA.c: (print_source_stats),
(print_stats), (main):
Add some example code for printing the RTP manager stats.
 2009-01-13 17:49:07 +00:00
Wim Taymans
996fb72681 tests/examples/rtp/server-decodebin-H263p-AMR.sh: Add example RTP transcoding pipeline from any file decodedable with... 
Original commit message from CVS:
* tests/examples/rtp/server-decodebin-H263p-AMR.sh:
Add example RTP transcoding pipeline from any file decodedable with
uridecodebin.
 2009-01-02 16:31:13 +00:00
Wim Taymans
9f46e70477 tests/examples/rtp/: Add two C examples of using gstrtpbin as a sender and a receiver. 
Original commit message from CVS:
* tests/examples/rtp/.cvsignore:
* tests/examples/rtp/Makefile.am:
* tests/examples/rtp/client-PCMA.c: (pad_added_cb), (main):
* tests/examples/rtp/server-alsasrc-PCMA.c: (main):
Add two C examples of using gstrtpbin as a sender and a receiver.
 2009-01-02 15:20:48 +00:00
Wim Taymans
a3dd91a4ae tests/examples/rtp/: Add some more H263p server and client examples. 
Original commit message from CVS:
* tests/examples/rtp/client-H263p.sdp:
* tests/examples/rtp/client-H263p.sh:
* tests/examples/rtp/server-VTS-H263p.sh:
Add some more H263p server and client examples.
 2008-10-07 09:58:13 +00:00
Wim Taymans
198224ef58 gst/rtsp/URLS: Some more urls. 
Original commit message from CVS:
* gst/rtsp/URLS:
Some more urls.
* gst/smpte/barboxwipes.c:
Add a comment
* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
Fix typo, add audioresample to the pipeline.
 2008-06-17 10:14:47 +00:00
Wim Taymans
81ec7c45ce tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Remove test sync-offset by default. 
Original commit message from CVS:
* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
Remove test sync-offset by default.
 2008-04-25 18:45:33 +00:00