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Document stuff.
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_clear_pt_map): * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (rtcp_thread), (gst_rtp_session_clear_pt_map): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init): Document stuff. Add clear-pt-map action signal where needed.
This commit is contained in:
Wim Taymans
parent
b2a310f5c0
commit
93888e03ac
14 changed files with 459 additions and 28 deletions
23
ChangeLog
23
ChangeLog
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@ -1,3 +1,26 @@
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2007-05-23 Wim Taymans <wim@fluendo.com>
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
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* docs/plugins/gst-plugins-bad-plugins-sections.txt:
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* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
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* gst/rtpmanager/gstrtpbin.h:
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* gst/rtpmanager/gstrtpclient.c:
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* gst/rtpmanager/gstrtpjitterbuffer.c:
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(gst_rtp_jitter_buffer_class_init),
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(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
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* gst/rtpmanager/gstrtpjitterbuffer.h:
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* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
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(gst_rtp_pt_demux_clear_pt_map):
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* gst/rtpmanager/gstrtpptdemux.h:
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* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
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(rtcp_thread), (gst_rtp_session_clear_pt_map):
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* gst/rtpmanager/gstrtpsession.h:
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* gst/rtpmanager/gstrtpssrcdemux.c:
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(gst_rtp_ssrc_demux_class_init):
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Document stuff.
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Add clear-pt-map action signal where needed.
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2007-05-22 Stefan Kost <ensonic@users.sf.net>
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* configure.ac:
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@ -105,6 +105,12 @@ EXTRA_HFILES = \
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$(top_srcdir)/gst/replaygain/gstrganalysis.h \
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$(top_srcdir)/gst/replaygain/gstrglimiter.h \
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$(top_srcdir)/gst/replaygain/gstrgvolume.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpbin.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpclient.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpjitterbuffer.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpptdemux.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpsession.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpssrcdemux.h \
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$(top_srcdir)/gst/videocrop/gstvideocrop.h
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# Images to copy into HTML directory.
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@ -22,6 +22,11 @@
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<xi:include href="xml/element-rganalysis.xml" />
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<xi:include href="xml/element-rglimiter.xml" />
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<xi:include href="xml/element-rgvolume.xml" />
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<xi:include href="xml/element-rtpbin.xml" />
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<xi:include href="xml/element-rtpjitterbuffer.xml" />
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<xi:include href="xml/element-rtpptdemux.xml" />
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<xi:include href="xml/element-rtpsession.xml" />
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<xi:include href="xml/element-rtpssrcdemux.xml" />
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<xi:include href="xml/element-sdlaudiosink.xml" />
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<xi:include href="xml/element-sdlvideosink.xml" />
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<xi:include href="xml/element-trm.xml" />
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@ -53,6 +58,7 @@
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<xi:include href="xml/plugin-osxvideo.xml" />
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<xi:include href="xml/plugin-qtdemux.xml" />
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<xi:include href="xml/plugin-replaygain.xml" />
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<xi:include href="xml/plugin-rtpmanager.xml" />
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<xi:include href="xml/plugin-sdl.xml" />
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<xi:include href="xml/plugin-spectrum.xml" />
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<xi:include href="xml/plugin-speed.xml" />
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@ -79,6 +79,96 @@ GstRgVolume
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GstRgVolumeClass
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</SECTION>
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<SECTION>
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<FILE>element-rtpbin</FILE>
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GstRTPBin
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<TITLE>rtpbin</TITLE>
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<SUBSECTION Standard>
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GstRTPBinPrivate
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GstRTPBinClass
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GST_RTP_BIN
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GST_IS_RTP_BIN
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GST_TYPE_RTP_BIN
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gst_rtp_bin_get_type
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GST_RTP_BIN_CLASS
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GST_IS_RTP_BIN_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-rtpclient</FILE>
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<TITLE>rtpclient</TITLE>
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GstRTPClient
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<SUBSECTION Standard>
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GstRTPClientClass
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GstRTPClientPrivate
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GST_RTP_CLIENT
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GST_IS_RTP_CLIENT
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GST_TYPE_RTP_CLIENT
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gst_rtp_client_get_type
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GST_RTP_CLIENT_CLASS
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GST_IS_RTP_CLIENT_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-rtpjitterbuffer</FILE>
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<TITLE>rtpjitterbuffer</TITLE>
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GstRTPJitterBuffer
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<SUBSECTION Standard>
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GstRTPJitterBufferClass
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GstRTPJitterBufferPrivate
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GST_RTP_JITTER_BUFFER
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GST_IS_RTP_JITTER_BUFFER
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GST_TYPE_RTP_JITTER_BUFFER
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gst_rtp_jitter_buffer_get_type
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GST_RTP_JITTER_BUFFER_CLASS
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GST_IS_RTP_JITTER_BUFFER_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-rtpptdemux</FILE>
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<TITLE>rtpptdemux</TITLE>
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GstRTPPtDemux
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<SUBSECTION Standard>
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GstRTPPtDemuxClass
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GstRTPPtDemuxPad
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GST_RTP_PT_DEMUX
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GST_IS_RTP_PT_DEMUX
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GST_TYPE_RTP_PT_DEMUX
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gst_rtp_pt_demux_get_type
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GST_RTP_PT_DEMUX_CLASS
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GST_IS_RTP_PT_DEMUX_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-rtpsession</FILE>
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<TITLE>rtpsession</TITLE>
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GstRTPSession
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<SUBSECTION Standard>
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GstRTPSessionClass
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GstRTPSessionPrivate
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GST_RTP_SESSION
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GST_IS_RTP_SESSION
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GST_TYPE_RTP_SESSION
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gst_rtp_session_get_type
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GST_RTP_SESSION_CLASS
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GST_IS_RTP_SESSION_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-rtpssrcdemux</FILE>
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<TITLE>rtpssrcdemux</TITLE>
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GstRTPSsrcDemux
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<SUBSECTION Standard>
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GstRTPSsrcDemuxClass
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GstRTPSsrcDemuxPad
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GST_RTP_SSRC_DEMUX
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GST_IS_RTP_SSRC_DEMUX
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GST_TYPE_RTP_SSRC_DEMUX
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gst_rtp_ssrc_demux_get_type
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GST_RTP_SSRC_DEMUX_CLASS
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GST_IS_RTP_SSRC_DEMUX_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-sdlaudiosink</FILE>
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GstSDLAudioSink
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@ -20,20 +20,64 @@
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/**
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* SECTION:element-rtpbin
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* @short_description: handle media from one RTP bin
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* @see_also: rtpjitterbuffer, rtpclient, rtpsession
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* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
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*
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* <refsect2>
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* <para>
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* RTP bin combines the functions of rtpsession, rtpssrcdemux, rtpjitterbuffer
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* and rtpptdemux in one element. It allows for multiple rtpsessions that will
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* be synchronized together using RTCP SR packets.
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* </para>
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* <para>
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* rtpbin is configured with a number of request pads that define the
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* functionality that is activated, similar to the rtpsession element.
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* </para>
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* <para>
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* To use rtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
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* number must be specified in the pad name.
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* Data received on the recv_rtp_sink_%%d pad will be processed in the rtpsession
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* manager and after being validated forwarded on rtpssrcdemuxer element. Each
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* RTP stream is demuxed based on the SSRC and send to a rtpjitterbuffer. After
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* the packets are released from the jitterbuffer, they will be forwarded to an
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* rtpptdemuxer element. The rtpptdemuxer element will demux the packets based
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* on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
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* rtpbin with the session number, SSRC and payload type respectively as the pad
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* name.
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* </para>
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* <para>
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* To also use rtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
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* session number must be specified in the pad name.
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* </para>
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* <para>
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* If you want the session manager to generate and send RTCP packets, request
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* the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed
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* on this pad contain SR/RR RTCP reports that should be sent to all participants
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* in the session.
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* </para>
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* <para>
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* To use rtpbin as a sender, request a send_rtp_sink_%%d pad, which will
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* automatically create a send_rtp_src_%%d pad. The session number must be specified when
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* requesting the sink pad. The session manager will modify the
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* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
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* send_rtp_src_%%d pad after updating its internal state.
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* </para>
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* <para>
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* The session manager needs the clock-rate of the payload types it is handling
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* and will signal the GstRTPSession::request-pt-map signal when it needs such a
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* mapping. One can clear the cached values with the GstRTPSession::clear-pt-map
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* signal.
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
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* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
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* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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* </programlisting>
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* Receive RTP data from port 5000 and send to the session 0 in rtpbin.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-04-02 (0.10.6)
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* Last reviewed on 2007-05-23 (0.10.6)
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*/
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#ifdef HAVE_CONFIG_H
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@ -50,7 +94,7 @@ GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
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/* elementfactory information */
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static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin",
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"Filter/Editor/Video",
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"Filter/Network/RTP",
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"Implement an RTP bin",
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"Wim Taymans <wim@fluendo.com>");
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@ -485,8 +529,8 @@ gst_rtp_bin_class_init (GstRTPBinClass * klass)
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g_object_class_install_property (gobject_class, PROP_LATENCY,
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g_param_spec_uint ("latency", "Buffer latency in ms",
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"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
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G_PARAM_READWRITE));
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"Default amount of ms to buffer in the jitterbuffers", 0,
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G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE));
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/**
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* GstRTPBin::request-pt-map:
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@ -501,10 +545,16 @@ gst_rtp_bin_class_init (GstRTPBinClass * klass)
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, request_pt_map),
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NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
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G_TYPE_UINT, G_TYPE_UINT);
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/**
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* GstRTPBin::clear-pt-map:
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* @rtpbin: the object which received the signal
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*
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* Clear all previously cached pt-mapping obtained with
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* GstRTPBin::request-pt-map.
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*/
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gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
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g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, clear_pt_map),
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G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTPBinClass, clear_pt_map),
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NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
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gstelement_class->provide_clock =
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@ -40,6 +40,7 @@ typedef struct _GstRTPBinPrivate GstRTPBinPrivate;
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struct _GstRTPBin {
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GstBin bin;
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/*< private >*/
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/* default latency for sessions */
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guint latency;
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/* a list of session */
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@ -51,7 +51,7 @@
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/* elementfactory information */
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static const GstElementDetails rtpclient_details =
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GST_ELEMENT_DETAILS ("RTP Client",
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"Filter/Editor/Video",
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"Filter/Network/RTP",
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"Implement an RTP client",
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"Wim Taymans <wim@fluendo.com>");
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@ -38,6 +38,15 @@
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* <para>
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* This element acts as a live element and so adds ::latency to the pipeline.
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* </para>
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* <para>
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* The element needs the clock-rate of the RTP payload in order to estimate the
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* delay. This information is obtained either from the caps on the sink pad or,
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* when no caps are present, from the ::request-pt-map signal. To clear the
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* previous pt-map use the ::clear-pt-map signal.
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* </para>
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* <para>
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* This element will automatically be used inside rtpbin.
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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@ -49,7 +58,7 @@
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-03-27 (0.10.13)
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* Last reviewed on 2007-05-22 (0.10.6)
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*/
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#ifdef HAVE_CONFIG_H
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@ -74,7 +83,7 @@ GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
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/* elementfactory information */
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static const GstElementDetails gst_rtp_jitter_buffer_details =
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GST_ELEMENT_DETAILS ("RTP packet jitter-buffer",
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"Filter/Network",
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"Filter/Network/RTP",
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"A buffer that deals with network jitter and other transmission faults",
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"Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
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"Wim Taymans <wim@fluendo.com>");
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@ -82,8 +91,8 @@ GST_ELEMENT_DETAILS ("RTP packet jitter-buffer",
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/* RTPJitterBuffer signals and args */
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enum
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{
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/* FILL ME */
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SIGNAL_REQUEST_PT_MAP,
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SIGNAL_CLEAR_PT_MAP,
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LAST_SIGNAL
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};
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@ -187,6 +196,9 @@ gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active);
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static void gst_rtp_jitter_buffer_loop (GstRTPJitterBuffer * jitterbuffer);
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static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query);
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static void
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gst_rtp_jitter_buffer_clear_pt_map (GstRTPJitterBuffer * jitterbuffer);
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static void
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gst_rtp_jitter_buffer_base_init (gpointer klass)
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{
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@ -215,17 +227,26 @@ gst_rtp_jitter_buffer_class_init (GstRTPJitterBufferClass * klass)
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gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
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gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
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/**
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* GstRTPJitterBuffer::latency:
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*
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* The maximum latency of the jitterbuffer. Packets will be kept in the buffer
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* for at most this time.
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*/
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g_object_class_install_property (gobject_class, PROP_LATENCY,
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g_param_spec_uint ("latency", "Buffer latency in ms",
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"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
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G_PARAM_READWRITE));
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/**
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* GstRTPJitterBuffer::drop-on-latency:
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*
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* Drop oldest buffers when the queue is completely filled.
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*/
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g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
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g_param_spec_boolean ("drop_on_latency",
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g_param_spec_boolean ("drop-on-latency",
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"Drop buffers when maximum latency is reached",
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"Tells the jitterbuffer to never exceed the given latency in size",
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DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE));
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/**
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* GstRTPJitterBuffer::request-pt-map:
|
||||
* @buffer: the object which received the signal
|
||||
|
|
@ -238,9 +259,22 @@ gst_rtp_jitter_buffer_class_init (GstRTPJitterBufferClass * klass)
|
|||
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPJitterBufferClass,
|
||||
request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
|
||||
GST_TYPE_CAPS, 1, G_TYPE_UINT);
|
||||
/**
|
||||
* GstRTPJitterBuffer::clear-pt-map:
|
||||
* @buffer: the object which received the signal
|
||||
*
|
||||
* Invalidate the clock-rate as obtained with the ::request-pt-map signal.
|
||||
*/
|
||||
gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
|
||||
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
|
||||
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPJitterBufferClass,
|
||||
clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID,
|
||||
G_TYPE_NONE, 0, G_TYPE_NONE);
|
||||
|
||||
gstelement_class->change_state = gst_rtp_jitter_buffer_change_state;
|
||||
|
||||
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT
|
||||
(rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
|
||||
}
|
||||
|
|
@ -305,6 +339,17 @@ gst_rtp_jitter_buffer_dispose (GObject * object)
|
|||
G_OBJECT_CLASS (parent_class)->dispose (object);
|
||||
}
|
||||
|
||||
static void
|
||||
gst_rtp_jitter_buffer_clear_pt_map (GstRTPJitterBuffer * jitterbuffer)
|
||||
{
|
||||
GstRTPJitterBufferPrivate *priv;
|
||||
|
||||
priv = jitterbuffer->priv;
|
||||
|
||||
/* this will trigger a new pt-map request signal, FIXME, do something better. */
|
||||
priv->clock_rate = -1;
|
||||
}
|
||||
|
||||
static GstCaps *
|
||||
gst_rtp_jitter_buffer_getcaps (GstPad * pad)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -49,13 +49,18 @@ typedef struct _GstRTPJitterBuffer GstRTPJitterBuffer;
|
|||
typedef struct _GstRTPJitterBufferClass GstRTPJitterBufferClass;
|
||||
typedef struct _GstRTPJitterBufferPrivate GstRTPJitterBufferPrivate;
|
||||
|
||||
/**
|
||||
* GstRTPJitterBuffer:
|
||||
*
|
||||
* Opaque jitterbuffer structure.
|
||||
*/
|
||||
struct _GstRTPJitterBuffer
|
||||
{
|
||||
GstElement parent;
|
||||
|
||||
/*< private >*/
|
||||
GstRTPJitterBufferPrivate *priv;
|
||||
|
||||
/*< private > */
|
||||
gpointer _gst_reserved[GST_PADDING];
|
||||
};
|
||||
|
||||
|
|
@ -66,6 +71,8 @@ struct _GstRTPJitterBufferClass
|
|||
/* signals */
|
||||
GstCaps* (*request_pt_map) (GstRTPJitterBuffer *buffer, guint pt);
|
||||
|
||||
void (*clear_pt_map) (GstRTPJitterBuffer *buffer);
|
||||
|
||||
/*< private > */
|
||||
gpointer _gst_reserved[GST_PADDING];
|
||||
};
|
||||
|
|
|
|||
|
|
@ -23,11 +23,42 @@
|
|||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
/**
|
||||
* SECTION:element-rtpptdemux
|
||||
* @short_description: separate RTP payloads based on the payload type
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
* rtpptdemux acts as a demuxer for RTP packets based on the payload type of the
|
||||
* packets. Its main purpose is to allow an application to easily receive and
|
||||
* decode an RTP stream with multiple payload types.
|
||||
* </para>
|
||||
* <para>
|
||||
* For each payload type that is detected, a new pad will be created and the
|
||||
* ::new-payload-type signal will be emitted. When the payload for the RTP
|
||||
* stream changes, the ::payload-type-change signal will be emitted.
|
||||
* </para>
|
||||
* <para>
|
||||
* The element will try to set complete and unique application/x-rtp caps on the
|
||||
* outgoing buffers and pads based on the result of the ::request-pt-map signal.
|
||||
* </para>
|
||||
* <title>Example pipelines</title>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink
|
||||
* </programlisting>
|
||||
* Takes an RTP stream and send the RTP packets with the first detected payload
|
||||
* type to fakesink, discarding the other payload types.
|
||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* Last reviewed on 2007-05-22 (0.10.6)
|
||||
*/
|
||||
|
||||
/*
|
||||
* Contributors:
|
||||
* Andre Moreira Magalhaes <andre.magalhaes@indt.org.br>
|
||||
*/
|
||||
|
||||
/*
|
||||
* Status:
|
||||
* - works with the test_rtpdemux.c tool
|
||||
|
|
@ -86,6 +117,7 @@ enum
|
|||
SIGNAL_REQUEST_PT_MAP,
|
||||
SIGNAL_NEW_PAYLOAD_TYPE,
|
||||
SIGNAL_PAYLOAD_TYPE_CHANGE,
|
||||
SIGNAL_CLEAR_PT_MAP,
|
||||
LAST_SIGNAL
|
||||
};
|
||||
|
||||
|
|
@ -99,6 +131,7 @@ static gboolean gst_rtp_pt_demux_setup (GstElement * element);
|
|||
static GstFlowReturn gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf);
|
||||
static GstStateChangeReturn gst_rtp_pt_demux_change_state (GstElement * element,
|
||||
GstStateChange transition);
|
||||
static void gst_rtp_pt_demux_clear_pt_map (GstRTPPtDemux * rtpdemux);
|
||||
|
||||
static GstPad *find_pad_for_pt (GstRTPPtDemux * rtpdemux, guint8 pt);
|
||||
|
||||
|
|
@ -106,8 +139,7 @@ static guint gst_rtp_pt_demux_signals[LAST_SIGNAL] = { 0 };
|
|||
|
||||
static GstElementDetails gst_rtp_pt_demux_details = {
|
||||
"RTP Demux",
|
||||
/* XXX: what's the correct hierarchy? */
|
||||
"Codec/Demux/Network",
|
||||
"Demux/Network/RTP",
|
||||
"Parses codec streams transmitted in the same RTP session",
|
||||
"Kai Vehmanen <kai.vehmanen@nokia.com>"
|
||||
};
|
||||
|
|
@ -148,7 +180,7 @@ gst_rtp_pt_demux_class_init (GstRTPPtDemuxClass * klass)
|
|||
G_TYPE_UINT);
|
||||
|
||||
/**
|
||||
* GstRTPPtDemux::new-payload-type
|
||||
* GstRTPPtDemux::new-payload-type:
|
||||
* @demux: the object which received the signal
|
||||
* @pt: the payload type
|
||||
* @pad: the pad with the new payload
|
||||
|
|
@ -162,7 +194,7 @@ gst_rtp_pt_demux_class_init (GstRTPPtDemuxClass * klass)
|
|||
G_TYPE_UINT, GST_TYPE_PAD);
|
||||
|
||||
/**
|
||||
* GstRTPPtDemux::payload-type-change
|
||||
* GstRTPPtDemux::payload-type-change:
|
||||
* @demux: the object which received the signal
|
||||
* @pt: the new payload type
|
||||
*
|
||||
|
|
@ -174,14 +206,28 @@ gst_rtp_pt_demux_class_init (GstRTPPtDemuxClass * klass)
|
|||
payload_type_change), NULL, NULL, g_cclosure_marshal_VOID__UINT,
|
||||
G_TYPE_NONE, 1, G_TYPE_UINT);
|
||||
|
||||
/**
|
||||
* GstRTPPtDemux::clear-pt-map:
|
||||
* @demux: the object which received the signal
|
||||
*
|
||||
* The application can call this signal to instruct the element to discard the
|
||||
* currently cached payload type map.
|
||||
*/
|
||||
gst_rtp_pt_demux_signals[SIGNAL_CLEAR_PT_MAP] =
|
||||
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
|
||||
G_SIGNAL_ACTION | G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPPtDemuxClass,
|
||||
clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID,
|
||||
G_TYPE_NONE, 0, G_TYPE_NONE);
|
||||
|
||||
gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_finalize);
|
||||
|
||||
gstelement_klass->change_state =
|
||||
GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_change_state);
|
||||
|
||||
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_clear_pt_map);
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT (gst_rtp_pt_demux_debug,
|
||||
"rtpptdemux", 0, "RTP codec demuxer");
|
||||
|
||||
}
|
||||
|
||||
static void
|
||||
|
|
@ -207,6 +253,12 @@ gst_rtp_pt_demux_finalize (GObject * object)
|
|||
G_OBJECT_CLASS (parent_class)->finalize (object);
|
||||
}
|
||||
|
||||
static void
|
||||
gst_rtp_pt_demux_clear_pt_map (GstRTPPtDemux * rtpdemux)
|
||||
{
|
||||
/* FIXME, do something */
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -53,6 +53,8 @@ struct _GstRTPPtDemuxClass
|
|||
|
||||
/* signal emitted when the payload type changes */
|
||||
void (*payload_type_change) (GstRTPPtDemux *demux, guint pt);
|
||||
|
||||
void (*clear_pt_map) (GstRTPPtDemux *demux);
|
||||
};
|
||||
|
||||
GType gst_rtp_pt_demux_get_type (void);
|
||||
|
|
|
|||
|
|
@ -20,20 +20,112 @@
|
|||
/**
|
||||
* SECTION:element-rtpsession
|
||||
* @short_description: an RTP session manager
|
||||
* @see_also: rtpjitterbuffer, rtpbin
|
||||
* @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
* The RTP session manager models one participant with a unique SSRC in an RTP
|
||||
* session. This session can be used to send and receive RTP and RTCP packets.
|
||||
* Based on what REQUEST pads are requested from the session manager, specific
|
||||
* functionality can be activated.
|
||||
* </para>
|
||||
* <para>
|
||||
* The session manager currently implements RFC 3550 including:
|
||||
* <itemizedlist>
|
||||
* <listitem>
|
||||
* <para>RTP packet validation based on consecutive sequence numbers.</para>
|
||||
* </listitem>
|
||||
* <listitem>
|
||||
* <para>Maintainance of the SSRC participant database.</para>
|
||||
* </listitem>
|
||||
* <listitem>
|
||||
* <para>Keeping per participant statistics based on received RTCP packets.</para>
|
||||
* </listitem>
|
||||
* <listitem>
|
||||
* <para>Scheduling of RR/SR RTCP packets.</para>
|
||||
* </listitem>
|
||||
* </itemizedlist>
|
||||
* </para>
|
||||
* <para>
|
||||
* The rtpsession will not demux packets based on SSRC or payload type, nor will
|
||||
* it correct for packet reordering and jitter. Use rtpssrcdemux, rtpptdemux and
|
||||
* rtpjitterbuffer in addition to rtpsession to perform these tasks. It is
|
||||
* usually a good idea to use rtpbin, which combines all these features in one
|
||||
* element.
|
||||
* </para>
|
||||
* <para>
|
||||
* To use rtpsession as an RTP receiver, request a recv_rtp_sink pad, which will
|
||||
* automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad
|
||||
* will be processed in the session and after being validated forwarded on the
|
||||
* recv_rtp_src pad.
|
||||
* </para>
|
||||
* <para>
|
||||
* To also use rtpsession as an RTCP receiver, request a recv_rtcp_sink pad,
|
||||
* which will automatically create a sync_src pad. Packets received on the RTCP
|
||||
* pad will be used by the session manager to update the stats and database of
|
||||
* the other participants. SR packets will be forwarded on the sync_src pad
|
||||
* so that they can be used to perform inter-stream synchronisation when needed.
|
||||
* </para>
|
||||
* <para>
|
||||
* If you want the session manager to generate and send RTCP packets, request
|
||||
* the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports
|
||||
* that should be sent to all participants in the session.
|
||||
* </para>
|
||||
* <para>
|
||||
* To use rtpsession as a sender, request a send_rtp_sink pad, which will
|
||||
* automatically create a send_rtp_src pad. The session manager will modify the
|
||||
* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
|
||||
* send_rtp_src pad after updating its internal state.
|
||||
* </para>
|
||||
* <para>
|
||||
* The session manager needs the clock-rate of the payload types it is handling
|
||||
* and will signal the GstRTPSession::request-pt-map signal when it needs such a
|
||||
* mapping. One can clear the cached values with the GstRTPSession::clear-pt-map
|
||||
* signal.
|
||||
* </para>
|
||||
* <title>Example pipelines</title>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
|
||||
* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
|
||||
* </programlisting>
|
||||
* Receive theora RTP packets from port 5000 and send them to the depayloader,
|
||||
* decoder and display. Note that the application/x-rtp caps on udpsrc should be
|
||||
* configured based on some negotiation process such as RTSP for this pipeline
|
||||
* to work correctly.
|
||||
* </para>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \
|
||||
* .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
|
||||
* udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
|
||||
* </programlisting>
|
||||
* Receive theora RTP packets from port 5000 and send them to the depayloader,
|
||||
* decoder and display. Receive RTCP packets from port 5001 and process them in
|
||||
* the session manager.
|
||||
* Note that the application/x-rtp caps on udpsrc should be
|
||||
* configured based on some negotiation process such as RTSP for this pipeline
|
||||
* to work correctly.
|
||||
* </para>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000
|
||||
* </programlisting>
|
||||
* Send theora RTP packets through the session manager and out on UDP port 5000.
|
||||
* </para>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \
|
||||
* ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
|
||||
* </programlisting>
|
||||
* Send theora RTP packets through the session manager and out on UDP port 5000.
|
||||
* Send RTCP packets on port 5001. Not that this pipeline will not preroll
|
||||
* correctly because the second udpsink will not preroll correctly (no RTCP
|
||||
* packets are sent in the PAUSED state). Applications should manually set and
|
||||
* keep (see #gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
|
||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* Last reviewed on 2007-04-02 (0.10.6)
|
||||
* Last reviewed on 2007-05-23 (0.10.6)
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
|
|
@ -50,7 +142,7 @@ GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
|
|||
/* elementfactory information */
|
||||
static const GstElementDetails rtpsession_details =
|
||||
GST_ELEMENT_DETAILS ("RTP Session",
|
||||
"Filter/Editor/Video",
|
||||
"Filter/Network/RTP",
|
||||
"Implement an RTP session",
|
||||
"Wim Taymans <wim@fluendo.com>");
|
||||
|
||||
|
|
@ -109,6 +201,7 @@ GST_STATIC_PAD_TEMPLATE ("send_rtcp_src",
|
|||
enum
|
||||
{
|
||||
SIGNAL_REQUEST_PT_MAP,
|
||||
SIGNAL_CLEAR_PT_MAP,
|
||||
LAST_SIGNAL
|
||||
};
|
||||
|
||||
|
|
@ -169,6 +262,8 @@ static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
|
|||
GstPadTemplate * templ, const gchar * name);
|
||||
static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
|
||||
|
||||
static void gst_rtp_session_clear_pt_map (GstRTPSession * rtpsession);
|
||||
|
||||
static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
|
||||
|
||||
GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
|
||||
|
|
@ -226,6 +321,16 @@ gst_rtp_session_class_init (GstRTPSessionClass * klass)
|
|||
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPSessionClass, request_pt_map),
|
||||
NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1,
|
||||
G_TYPE_UINT);
|
||||
/**
|
||||
* GstRTPSession::clear-pt-map:
|
||||
* @sess: the object which received the signal
|
||||
*
|
||||
* Clear the cached pt-maps requested with GstRTPSession::request-pt-map.
|
||||
*/
|
||||
gst_rtp_session_signals[SIGNAL_CLEAR_PT_MAP] =
|
||||
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
|
||||
G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTPSessionClass, clear_pt_map),
|
||||
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
|
||||
|
||||
gstelement_class->change_state =
|
||||
GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
|
||||
|
|
@ -234,6 +339,8 @@ gst_rtp_session_class_init (GstRTPSessionClass * klass)
|
|||
gstelement_class->release_pad =
|
||||
GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
|
||||
|
||||
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
|
||||
"rtpsession", 0, "RTP Session");
|
||||
}
|
||||
|
|
@ -315,7 +422,6 @@ rtcp_thread (GstRTPSession * rtpsession)
|
|||
next_timeout =
|
||||
rtp_session_next_timeout (rtpsession->priv->session, current_time);
|
||||
|
||||
|
||||
GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (next_timeout));
|
||||
|
||||
|
|
@ -438,6 +544,12 @@ failed_thread:
|
|||
}
|
||||
}
|
||||
|
||||
static void
|
||||
gst_rtp_session_clear_pt_map (GstRTPSession * rtpsession)
|
||||
{
|
||||
/* FIXME, do something */
|
||||
}
|
||||
|
||||
/* called when the session manager has an RTP packet ready for further
|
||||
* processing */
|
||||
static GstFlowReturn
|
||||
|
|
|
|||
|
|
@ -59,6 +59,8 @@ struct _GstRTPSessionClass {
|
|||
|
||||
/* signals */
|
||||
GstCaps* (*request_pt_map) (GstRTPSession *sess, guint pt);
|
||||
|
||||
void (*clear_pt_map) (GstRTPSession *sess);
|
||||
};
|
||||
|
||||
GType gst_rtp_session_get_type (void);
|
||||
|
|
|
|||
|
|
@ -19,6 +19,33 @@
|
|||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
/**
|
||||
* SECTION:element-rtpssrcdemux
|
||||
* @short_description: separate RTP payloads based on the SSRC
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
* rtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
|
||||
* packets. Its main purpose is to allow an application to easily receive and
|
||||
* decode an RTP stream with multiple SSRCs.
|
||||
* </para>
|
||||
* <para>
|
||||
* For each SSRC that is detected, a new pad will be created and the
|
||||
* ::new-ssrc-pad signal will be emitted.
|
||||
* </para>
|
||||
* <title>Example pipelines</title>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink
|
||||
* </programlisting>
|
||||
* Takes an RTP stream and send the RTP packets with the first detected SSRC
|
||||
* to fakesink, discarding the other SSRCs.
|
||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* Last reviewed on 2007-05-23 (0.10.6)
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include "config.h"
|
||||
#endif
|
||||
|
|
@ -49,7 +76,7 @@ GST_STATIC_PAD_TEMPLATE ("src_%d",
|
|||
|
||||
static GstElementDetails gst_rtp_ssrc_demux_details = {
|
||||
"RTP SSRC Demux",
|
||||
"Codec/Demux/Network",
|
||||
"Demux/Network/RTP",
|
||||
"Splits RTP streams based on the SSRC",
|
||||
"Wim Taymans <wim@fluendo.com>"
|
||||
};
|
||||
|
|
@ -165,6 +192,14 @@ gst_rtp_ssrc_demux_class_init (GstRTPSsrcDemuxClass * klass)
|
|||
|
||||
gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_finalize);
|
||||
|
||||
/**
|
||||
* GstRTPSsrcDemux::new-ssrc-pad:
|
||||
* @demux: the object which received the signal
|
||||
* @ssrc: the SSRC of the pad
|
||||
* @pad: the new pad.
|
||||
*
|
||||
* Emited when a new SSRC pad has been created.
|
||||
*/
|
||||
gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD] =
|
||||
g_signal_new ("new-ssrc-pad",
|
||||
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
|
||||
|
|
|
|||
Loading…
Reference in a new issue