This is a re-implementation of the RTP elements that are submitted in
2013 to handle RTP streams. The elements handle a correct connection
for the bi-directional use of the RTCP sockets.
https://bugzilla.gnome.org/show_bug.cgi?id=703111
The rtpsink and rtpsrc elements add an URI interface so that streams
can be decoded with decodebin using the rtp:// interface.
The code can be used as follows
```
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink
gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink
```
rtpmanagerbad: add pkg-config
rtpmanagerbad: Rtp should be uppercase
rtpmanagerbad: add G_OS_WIN32 for shielding unix headers
rtpmanagerbad: remove Since from documentation
rtpmanagerbad: rename lib name from nrtp to rtpmanagerbad
rtpmanagerbad: sync meson.build with other modules
rtpmanagerbad: add Makefile.am
rtpmanagerbad: use GstElement to count pads
rtpmanagerbad: use gst_bin_set_suppressed_flags
rtpmanagerbad: check element creation
rtpmanagerbad: post message when trying to access missing rtpbin
rtpmanagerbad: return FALSE with g_return tests
rtpmanagerbad: use gsocket multicast check
rtpmanagerbad: use gst_caps_new_empty_simple iso gst_caps_from_string
rtpmanagerbad: sync with gstrtppayloads.h
rtpmanagerbad: correct media type X-GST
rtpmanagerbad: test if a compatible pad was found
rtpmanagerbad: remove evil copy of GstRTPPayloadInfo
rtpmanagerbad: add gio_dep to meson
rtpmanagerbad: revert to old glib boilerplate
GStreamer 1.16 does not yet support the newer GLib templates, so revert.
rtpmanagerbad: return GST_STATE_CHANGE_NO_PREROLL for live sources
for live sources, NO_PREROLL should be returned for PLAYING->PAUSED and
READY->PAUSED transitions.
rtpmanagerbad: use GstElement pad counting
rtpmanagerbad: just use template name to request pad
rtpmanagerbad: remove commented code
rtpmanagerbad: use funnel to send multiple streams on one socket
rtpmanagerbad: avoid beaches
beaches should only be used during the summer, so rewrite the code to
return explicitly and avoid beaches during the winter.
rtpmanagerbad: add copyright to test code
rtpmanagerbad: g_free is NULL safe
rtpmanagerbad: do not trace rtpbin
rtpmanagerbad: return NULL explitly
rtpmanagerbad: warn when data port is not even
According to RFC 3550, RTP data should be sent on even ports, while RTCP
is sent on the following odd port.
rtpmanagerbad: document port allocation in rtpsink/src
rtpmanagerbad: improve uri description
rtpmanagerbad: add comment re-use socket
rtpmanagerbad: rename gst_object_set_properties_from_uri_query
rtpmanagerbad: loan prop/val setter from rist
rtpmanagerbad: rtpsrc: fix unitialised pointer
rtpmanagerbad: fix silly typo
rtpmanagerbad: test for empty key/value
rtpmanagerbad: rtpsrc: deprecate ssrc collision to INFO
rtpmanagerbad: sync debug with rist
rtpmanagerbad: small strings allocated on stack
rtpmanagerbad: correct rename
rtpmanagerbad: add locking on prop setters/getters
Locking is added because the URI allows to access the properties too.
rtpmanagerbad: allow for RTCP through NAT
rtpmanagerbad: move gio to header file
rtpmanagerbad: free small strings too
rtpmanagerbad: ttl_mc for ttl on dynudpsink
rtpmanagerbad: add comments on the URI registered
rtpmanagerbad: correct macro after file rename
rtpmanagerbad: code style
rtpmanagerbad: handle wrong URIs in setter
rtpmanagerbad: nit URI notation correction
In an URI, the first key/value pair should not have an ampersand, the
parser did not die though.
Based upon the souphttpsrc tests, add unit tests for the curlhttpsrc
element. The souphttpsrc tests are able to use an HTTP server that
is provided as part of the soup library. This does not exist in the
curl library, therefore these tests provide a very simple HTTP server
using the GIO library.
These curlhttpsrc tests contain one new test that does not come from
the souphttpsrc tests. The test_multiple_http_requests test tries to
reproduce the way in which GstAdaptiveDemux makes use of URI source
elements. GstAdaptiveDemux creates a bin with the httpsrc element
and a queue element and sets the locked state of that bin to TRUE,
so that it does not follow the state transitions of its parent. It
then moves this bin to the PLAYING state to start each download and
back to READY when the download completes.
The VCD source was ported in 2014 (commit 89eb1e9), but the necessary
"cdxaparse" plugin, which is used to "Parse a .dat file (VCD) into
raw mpeg1" was never ported.
This means that the probable main user for the feature, totem, hasn't
actually been able to play back VCDs, since 2012, when it switched to
using GStreamer 1.0.
Note that even if cdxaparse was finally ported, a lot of work would
still be necessary before it is considered usable. Notably, it is
missing disc image support [1] and some VCDs just cannot be opened for
reading [2].
[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/898
[2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/899
This is for the same reason as the dash tests. This should ideally
be converted to gst-validate tests. These tests randomly timeout also
due to the tests doing seeks from the streaming thread (sic).
It works like a valve in front of the actual avwait. When recording ==
TRUE, other rules are then examined. When recording == FALSE, nothing is
passing through.
https://bugzilla.gnome.org/show_bug.cgi?id=796836
We used to have the same enum to represent H265 profiles and idc values.
Those are no longer the same with extension profiles defined from
version 2 of the spec.
Split those enums so the semantic of each is clearer and we'll be able
to add extension profiles to GstH265Profile.
Also add gst_h265_profile_tier_level_get_profile() to retrieve the
GstH265Profile from the GstH265ProfileTierLevel. It will be used to
implement the detection of extension profiles.
https://bugzilla.gnome.org/show_bug.cgi?id=793876
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
Except for gst/gl/gstglfuncs.h
It is up to the client app to include these headers.
It is coherent with the fact that gstreamer-gl.pc does not
require any egl.pc/gles.pc. I.e. it is the responsability
of the app to search these headers within its build setup.
For example gstreamer-vaapi includes explicitly EGL/egl.h
and search for it in its configure.ac.
For example with this patch, if an app includes the headers
gst/gl/egl/gstglcontext_egl.h
gst/gl/egl/gstgldisplay_egl.h
gst/gl/egl/gstglmemoryegl.h
it will *no longer* automatically include EGL/egl.h and GLES2/gl2.h.
Which is good because the app might want to use the gstgl api only
without the need to bother about gl headers.
Also added a test: cd tests/check && make libs/gstglheaders.check
https://bugzilla.gnome.org/show_bug.cgi?id=784779
If they were not ported after 4+ years it seems unlikely that anybody is
ever going to need them again. They're still in the GIT history if
needed.
https://bugzilla.gnome.org/show_bug.cgi?id=774530
This was used by MSN messenger in prehistoric times, it's safe
to say no one needs or wants this any more these days. For
decoding old recordings there's still a decoder in ffmpeg.
https://bugzilla.gnome.org/show_bug.cgi?id=597616
pnm: add misssing continuation slash at end of line
vp8parser: no need to link codecparsers twice
dash_demux, mssdemux, hlsdemux: fix lib/flags ordering - local
libs and flags must come first, then base before core before rest
External xml could have empty, one or multiple top-level "Period" elements.
Because xml parser cannot parse the multiple top-level elements
(i.e., no root element), we need to wrap a xml in order to make root element.
See also ISO/IEC 23009-1:2014 5.3.2.2
https://bugzilla.gnome.org/show_bug.cgi?id=774357
The linker flags for several plugins (dashdemux, mssdemux, hlsdemux) are missing
$(GST_PLUGINS_BASE_LIBS), so add them so they can find -lgsttag and -lgstapp.
https://bugzilla.gnome.org/show_bug.cgi?id=771780
It uses random data streams but dashdemux nowadays actually looks into the
streams and doesn't like randomness very much. The tests should probably just
become validate tests on real streams.
https://bugzilla.gnome.org/show_bug.cgi?id=769553
