diff --git a/configure.ac b/configure.ac index 22ce577031..3b34fa2207 100644 --- a/configure.ac +++ b/configure.ac @@ -302,6 +302,7 @@ AG_GST_CHECK_PLUGIN(multipart) AG_GST_CHECK_PLUGIN(qtdemux) AG_GST_CHECK_PLUGIN(replaygain) AG_GST_CHECK_PLUGIN(rtp) +AG_GST_CHECK_PLUGIN(rtpmanager) AG_GST_CHECK_PLUGIN(rtsp) AG_GST_CHECK_PLUGIN(smpte) AG_GST_CHECK_PLUGIN(spectrum) @@ -1065,6 +1066,7 @@ gst/multipart/Makefile gst/qtdemux/Makefile gst/replaygain/Makefile gst/rtp/Makefile +gst/rtpmanager/Makefile gst/rtsp/Makefile gst/smpte/Makefile gst/spectrum/Makefile diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index dd189640a2..3175dd1b5f 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -180,6 +180,11 @@ EXTRA_HFILES = \ $(top_srcdir)/gst/replaygain/gstrglimiter.h \ $(top_srcdir)/gst/replaygain/gstrgvolume.h \ $(top_srcdir)/gst/rtp/gstrtpjpegpay.h \ + $(top_srcdir)/gst/rtpmanager/gstrtpbin.h \ + $(top_srcdir)/gst/rtpmanager/gstrtpjitterbuffer.h \ + $(top_srcdir)/gst/rtpmanager/gstrtpptdemux.h \ + $(top_srcdir)/gst/rtpmanager/gstrtpsession.h \ + $(top_srcdir)/gst/rtpmanager/gstrtpssrcdemux.h \ $(top_srcdir)/gst/rtsp/gstrtpdec.h \ $(top_srcdir)/gst/rtsp/gstrtspsrc.h \ $(top_srcdir)/gst/smpte/gstsmpte.h \ diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml index 75714e3eed..742f0954a0 100644 --- a/docs/plugins/gst-plugins-good-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml @@ -77,6 +77,11 @@ + + + + + @@ -204,6 +209,7 @@ + diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index 73d7aff2ad..7f85d55dd4 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -843,6 +843,82 @@ GST_GOOM_CLASS GST_IS_GOOM_CLASS +
+element-gstrtpbin +gstrtpbin +GstRtpBin + +GstRtpBinPrivate +GstRtpBinClass +GST_RTP_BIN +GST_IS_RTP_BIN +GST_TYPE_RTP_BIN +gst_rtp_bin_get_type +GST_RTP_BIN_CLASS +GST_IS_RTP_BIN_CLASS +
+ +
+element-gstrtpjitterbuffer +gstrtpjitterbuffer +GstRtpJitterBuffer + +GstRtpJitterBufferClass +GstRtpJitterBufferPrivate +GST_RTP_JITTER_BUFFER +GST_IS_RTP_JITTER_BUFFER +GST_TYPE_RTP_JITTER_BUFFER +gst_rtp_jitter_buffer_get_type +GST_RTP_JITTER_BUFFER_CLASS +GST_IS_RTP_JITTER_BUFFER_CLASS +
+ +
+element-gstrtpptdemux +gstrtpptdemux +GstRtpPtDemux + +GstRtpPtDemuxClass +GstRtpPtDemuxPad +GST_RTP_PT_DEMUX +GST_IS_RTP_PT_DEMUX +GST_TYPE_RTP_PT_DEMUX +gst_rtp_pt_demux_get_type +GST_RTP_PT_DEMUX_CLASS +GST_IS_RTP_PT_DEMUX_CLASS +
+ +
+element-gstrtpsession +gstrtpsession +GstRtpSession + +GstRtpSessionClass +GstRtpSessionPrivate +GST_RTP_SESSION +GST_IS_RTP_SESSION +GST_TYPE_RTP_SESSION +gst_rtp_session_get_type +GST_RTP_SESSION_CLASS +GST_IS_RTP_SESSION_CLASS +GST_RTP_SESSION_CAST +
+ +
+element-gstrtpssrcdemux +gstrtpssrcdemux +GstRtpSsrcDemux + +GstRtpSsrcDemuxClass +GstRtpSsrcDemuxPad +GST_RTP_SSRC_DEMUX +GST_IS_RTP_SSRC_DEMUX +GST_TYPE_RTP_SSRC_DEMUX +gst_rtp_ssrc_demux_get_type +GST_RTP_SSRC_DEMUX_CLASS +GST_IS_RTP_SSRC_DEMUX_CLASS +
+
element-halaudiosink halaudiosink diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml new file mode 100644 index 0000000000..377f1d19af --- /dev/null +++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml @@ -0,0 +1,190 @@ + + gstrtpmanager + RTP session management plugin library + ../../gst/rtpmanager/.libs/libgstrtpmanager.so + libgstrtpmanager.so + 0.10.15.1 + LGPL + gst-plugins-good + GStreamer Good Plug-ins git/prerelease + Unknown package origin + + + gstrtpbin + RTP Bin + Filter/Network/RTP + Implement an RTP bin + Wim Taymans <wim.taymans@gmail.com> + + + send_rtp_src_%d + source + sometimes +
application/x-rtp
+ + + send_rtcp_src_%d + source + request +
application/x-rtcp
+ + + recv_rtp_src_%d_%d_%d + source + sometimes +
application/x-rtp
+ + + send_rtp_sink_%d + sink + request +
application/x-rtp
+ + + recv_rtcp_sink_%d + sink + request +
application/x-rtcp
+ + + recv_rtp_sink_%d + sink + request +
application/x-rtp
+ + + + + gstrtpjitterbuffer + RTP packet jitter-buffer + Filter/Network/RTP + A buffer that deals with network jitter and other transmission faults + Philippe Kalaf <philippe.kalaf@collabora.co.uk>, Wim Taymans <wim.taymans@gmail.com> + + + sink_rtcp + sink + request +
application/x-rtcp
+ + + sink + sink + always +
application/x-rtp, clock-rate=(int)[ 1, 2147483647 ]
+ + + src + source + always +
application/x-rtp
+ + + + + gstrtpptdemux + RTP Demux + Demux/Network/RTP + Parses codec streams transmitted in the same RTP session + Kai Vehmanen <kai.vehmanen@nokia.com> + + + src_%d + source + sometimes +
application/x-rtp, payload=(int)[ 0, 255 ]
+ + + sink + sink + always +
application/x-rtp
+ + + + + gstrtpsession + RTP Session + Filter/Network/RTP + Implement an RTP session + Wim Taymans <wim.taymans@gmail.com> + + + send_rtcp_src + source + request +
application/x-rtcp
+ + + send_rtp_src + source + sometimes +
application/x-rtp
+ + + sync_src + source + sometimes +
application/x-rtcp
+ + + recv_rtp_src + source + sometimes +
application/x-rtp
+ + + send_rtp_sink + sink + request +
application/x-rtp
+ + + recv_rtcp_sink + sink + request +
application/x-rtcp
+ + + recv_rtp_sink + sink + request +
application/x-rtp
+ + + + + gstrtpssrcdemux + RTP SSRC Demux + Demux/Network/RTP + Splits RTP streams based on the SSRC + Wim Taymans <wim.taymans@gmail.com> + + + rtcp_src_%d + source + sometimes +
application/x-rtcp
+ + + src_%d + source + sometimes +
application/x-rtp
+ + + rtcp_sink + sink + always +
application/x-rtcp
+ + + sink + sink + always +
application/x-rtp
+ + + + + \ No newline at end of file diff --git a/gst-plugins-good.spec.in b/gst-plugins-good.spec.in index 9e55ba0b5b..2586c7b180 100644 --- a/gst-plugins-good.spec.in +++ b/gst-plugins-good.spec.in @@ -100,6 +100,7 @@ rm -rf $RPM_BUILD_ROOT %{_libdir}/gstreamer-%{majorminor}/libgstmulaw.so %{_libdir}/gstreamer-%{majorminor}/libgstqtdemux.so %{_libdir}/gstreamer-%{majorminor}/libgstrtp.so +%{_libdir}/gstreamer-%{majorminor}/libgstrtpmanager.so %{_libdir}/gstreamer-%{majorminor}/libgstrtsp.so %{_libdir}/gstreamer-%{majorminor}/libgstsmpte.so %{_libdir}/gstreamer-%{majorminor}/libgstudp.so diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index d5895314dd..81c748c1fc 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -112,6 +112,8 @@ check_PROGRAMS = \ elements/rglimiter \ elements/rgvolume \ elements/rtp-payloading \ + elements/rtpbin \ + elements/rtpbin_buffer_list \ elements/spectrum \ elements/udpsink \ elements/videocrop \ @@ -169,6 +171,13 @@ elements_deinterleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMIN elements_interleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS) elements_interleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(LDADD) +elements_rtpbin_buffer_list_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \ + $(ERROR_CFLAGS) $(GST_CHECK_CFLAGS) +elements_rtpbin_buffer_list_LDADD = $(GST_PLUGINS_BASE_LIBS) \ + -lgstnetbuffer-@GST_MAJORMINOR@ -lgstrtp-@GST_MAJORMINOR@ \ + $(GST_BASE_LIBS) $(GST_LIBS_LIBS) $(GST_CHECK_LIBS) +elements_rtpbin_buffer_list_SOURCES = elements/rtpbin_buffer_list.c + elements_souphttpsrc_CFLAGS = $(SOUP_CFLAGS) $(AM_CFLAGS) elements_souphttpsrc_LDADD = $(SOUP_LIBS) $(LDADD) diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore index 88218ae714..a9ff8af85a 100644 --- a/tests/check/elements/.gitignore +++ b/tests/check/elements/.gitignore @@ -34,6 +34,8 @@ rganalysis rglimiter rgvolume rtp-payloading +rtpbin +rtpbin_buffer_list souphttpsrc spectrum sunaudio diff --git a/tests/check/pipelines/.gitignore b/tests/check/pipelines/.gitignore index 8513231a01..3b499050b9 100644 --- a/tests/check/pipelines/.gitignore +++ b/tests/check/pipelines/.gitignore @@ -1,4 +1,5 @@ .dirstamp +effectv flacdec simple-launch-lines wavpack