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fcce4aff92
Original commit message from CVS: * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream), (gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp): * gst/rtpmanager/gstrtpbin.h: Updated example pipelines in docs. Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync. Set the default latency correctly. Add some more points where we can get caps. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Add ts-offset property to control timestamping. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (get_current_ntp_ns_time), (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp), (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Various cleanups. Feed rtpsession manager with NTP time based on pipeline clock when handling RTP packets and RTCP timeouts. Perform all RTCP with the system clock. Set caps on RTCP outgoing buffers. * gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc), (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain): * gst/rtpmanager/gstrtpssrcdemux.h: Also demux RTCP messages. * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_rb), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_send_bye), (session_start_rtcp), (session_report_blocks), (session_cleanup), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Remove the get_time callback, the GStreamer part will feed us with enough timing information. Split sync timing and RTCP timing information. Factor out common RB handling for SR and RR. Send out SR RTCP packets for lip-sync. Move SR and RR packet info generation to the source. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb), (rtp_source_get_new_sr), (rtp_source_get_new_rb), (rtp_source_get_last_sr): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Use caps on incomming buffers to get timing information when they are there. Calculate clock scew of the receiver compared to the sender and adjust the rtp timestamps. Calculate the round trip in sources. Do SR and RR calculations in the source.
512 lines
14 KiB
C
512 lines
14 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
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*
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* RTP SSRC demuxer
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-gstrtpssrcdemux
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* @short_description: separate RTP payloads based on the SSRC
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*
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* <refsect2>
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* <para>
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* gstrtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
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* packets. Its main purpose is to allow an application to easily receive and
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* decode an RTP stream with multiple SSRCs.
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* </para>
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* <para>
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* For each SSRC that is detected, a new pad will be created and the
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* ::new-ssrc-pad signal will be emitted.
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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 udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
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* </programlisting>
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* Takes an RTP stream and send the RTP packets with the first detected SSRC
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* to fakesink, discarding the other SSRCs.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-05-28 (0.10.5)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include "gstrtpbin-marshal.h"
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#include "gstrtpssrcdemux.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_ssrc_demux_debug);
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#define GST_CAT_DEFAULT gst_rtp_ssrc_demux_debug
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/* generic templates */
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static GstStaticPadTemplate rtp_ssrc_demux_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtp_ssrc_demux_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtp_ssrc_demux_src_template =
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GST_STATIC_PAD_TEMPLATE ("src_%d",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtp_ssrc_demux_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_src_%d",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstElementDetails gst_rtp_ssrc_demux_details = {
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"RTP SSRC Demux",
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"Demux/Network/RTP",
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"Splits RTP streams based on the SSRC",
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"Wim Taymans <wim@fluendo.com>"
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};
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/* signals */
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enum
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{
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SIGNAL_NEW_SSRC_PAD,
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LAST_SIGNAL
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};
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GST_BOILERPLATE (GstRtpSsrcDemux, gst_rtp_ssrc_demux, GstElement,
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GST_TYPE_ELEMENT);
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/* GObject vmethods */
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static void gst_rtp_ssrc_demux_finalize (GObject * object);
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/* GstElement vmethods */
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static GstStateChangeReturn gst_rtp_ssrc_demux_change_state (GstElement *
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element, GstStateChange transition);
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/* sinkpad stuff */
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static GstFlowReturn gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad,
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GstBuffer * buf);
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static gboolean gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad,
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GstEvent * event);
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/* srcpad stuff */
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static gboolean gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event);
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static guint gst_rtp_ssrc_demux_signals[LAST_SIGNAL] = { 0 };
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/**
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* Item for storing GstPad <-> SSRC pairs.
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*/
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struct _GstRtpSsrcDemuxPad
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{
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guint32 ssrc;
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GstPad *rtp_pad;
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GstCaps *caps;
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GstPad *rtcp_pad;
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};
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/* find a src pad for a given SSRC, returns NULL if the SSRC was not found
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*/
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static GstRtpSsrcDemuxPad *
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find_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc)
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{
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GSList *walk;
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for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
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GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
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if (pad->ssrc == ssrc)
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return pad;
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}
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return NULL;
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}
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static GstRtpSsrcDemuxPad *
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create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc)
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{
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GstPad *rtp_pad, *rtcp_pad;
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GstElementClass *klass;
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GstPadTemplate *templ;
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gchar *padname;
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GstRtpSsrcDemuxPad *demuxpad;
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GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
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klass = GST_ELEMENT_GET_CLASS (demux);
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templ = gst_element_class_get_pad_template (klass, "src_%d");
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padname = g_strdup_printf ("src_%d", ssrc);
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rtp_pad = gst_pad_new_from_template (templ, padname);
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g_free (padname);
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templ = gst_element_class_get_pad_template (klass, "rtcp_src_%d");
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padname = g_strdup_printf ("rtcp_src_%d", ssrc);
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rtcp_pad = gst_pad_new_from_template (templ, padname);
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g_free (padname);
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/* wrap in structure and add to list */
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demuxpad = g_new0 (GstRtpSsrcDemuxPad, 1);
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demuxpad->ssrc = ssrc;
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demuxpad->rtp_pad = rtp_pad;
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demuxpad->rtcp_pad = rtcp_pad;
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demux->srcpads = g_slist_prepend (demux->srcpads, demuxpad);
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GST_OBJECT_UNLOCK (demux);
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/* copy caps from input */
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gst_pad_set_caps (rtp_pad, GST_PAD_CAPS (demux->rtp_sink));
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gst_pad_use_fixed_caps (rtp_pad);
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gst_pad_set_caps (rtcp_pad, GST_PAD_CAPS (demux->rtcp_sink));
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gst_pad_use_fixed_caps (rtcp_pad);
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gst_pad_set_event_function (rtp_pad, gst_rtp_ssrc_demux_src_event);
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gst_pad_set_active (rtp_pad, TRUE);
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gst_pad_set_active (rtcp_pad, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), rtp_pad);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), rtcp_pad);
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g_signal_emit (G_OBJECT (demux),
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gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD], 0, ssrc, rtp_pad);
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GST_OBJECT_LOCK (demux);
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return demuxpad;
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}
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static void
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gst_rtp_ssrc_demux_base_init (gpointer g_class)
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{
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GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_sink_template));
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_sink_template));
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_src_template));
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_src_template));
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gst_element_class_set_details (gstelement_klass, &gst_rtp_ssrc_demux_details);
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}
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static void
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gst_rtp_ssrc_demux_class_init (GstRtpSsrcDemuxClass * klass)
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{
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GObjectClass *gobject_klass;
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GstElementClass *gstelement_klass;
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gobject_klass = (GObjectClass *) klass;
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gstelement_klass = (GstElementClass *) klass;
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gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_finalize);
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/**
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* GstRtpSsrcDemux::new-ssrc-pad:
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* @demux: the object which received the signal
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* @ssrc: the SSRC of the pad
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* @pad: the new pad.
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*
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* Emited when a new SSRC pad has been created.
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*/
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gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD] =
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g_signal_new ("new-ssrc-pad",
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G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET (GstRtpSsrcDemuxClass, new_ssrc_pad),
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NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_OBJECT,
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G_TYPE_NONE, 2, G_TYPE_UINT, GST_TYPE_PAD);
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gstelement_klass->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_change_state);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_ssrc_demux_debug,
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"rtpssrcdemux", 0, "RTP SSRC demuxer");
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}
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static void
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gst_rtp_ssrc_demux_init (GstRtpSsrcDemux * demux,
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GstRtpSsrcDemuxClass * g_class)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (demux);
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demux->rtp_sink =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"sink"), "sink");
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gst_pad_set_chain_function (demux->rtp_sink, gst_rtp_ssrc_demux_chain);
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gst_pad_set_event_function (demux->rtp_sink, gst_rtp_ssrc_demux_sink_event);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtp_sink);
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demux->rtcp_sink =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"rtcp_sink"), "rtcp_sink");
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gst_pad_set_chain_function (demux->rtcp_sink, gst_rtp_ssrc_demux_rtcp_chain);
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gst_pad_set_event_function (demux->rtcp_sink,
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gst_rtp_ssrc_demux_rtcp_sink_event);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtcp_sink);
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}
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static void
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gst_rtp_ssrc_demux_finalize (GObject * object)
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{
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GstRtpSsrcDemux *demux;
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demux = GST_RTP_SSRC_DEMUX (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event)
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{
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GstRtpSsrcDemux *demux;
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gboolean res = FALSE;
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demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_NEWSEGMENT:
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default:
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{
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GSList *walk;
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res = TRUE;
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GST_OBJECT_LOCK (demux);
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for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
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GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
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gst_event_ref (event);
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res &= gst_pad_push_event (pad->rtp_pad, event);
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}
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GST_OBJECT_UNLOCK (demux);
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gst_event_unref (event);
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break;
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}
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}
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gst_object_unref (demux);
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return res;
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}
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static gboolean
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gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad, GstEvent * event)
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{
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GstRtpSsrcDemux *demux;
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gboolean res = FALSE;
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demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_NEWSEGMENT:
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default:
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{
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GSList *walk;
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res = TRUE;
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GST_OBJECT_LOCK (demux);
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for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
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GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
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res &= gst_pad_push_event (pad->rtcp_pad, event);
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}
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GST_OBJECT_UNLOCK (demux);
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break;
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}
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}
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gst_object_unref (demux);
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return res;
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}
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static GstFlowReturn
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gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf)
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{
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GstFlowReturn ret;
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GstRtpSsrcDemux *demux;
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guint32 ssrc;
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GstRtpSsrcDemuxPad *dpad;
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demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
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if (!gst_rtp_buffer_validate (buf))
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goto invalid_payload;
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ssrc = gst_rtp_buffer_get_ssrc (buf);
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GST_DEBUG_OBJECT (demux, "received buffer of SSRC %08x", ssrc);
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GST_OBJECT_LOCK (demux);
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dpad = find_demux_pad_for_ssrc (demux, ssrc);
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if (dpad == NULL) {
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if (!(dpad = create_demux_pad_for_ssrc (demux, ssrc)))
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goto create_failed;
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}
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GST_OBJECT_UNLOCK (demux);
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/* push to srcpad */
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ret = gst_pad_push (dpad->rtp_pad, buf);
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return ret;
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/* ERRORS */
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invalid_payload:
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{
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/* this is fatal and should be filtered earlier */
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GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Dropping invalid RTP payload"));
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
create_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Could not create new pad"));
|
|
GST_OBJECT_UNLOCK (demux);
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstRtpSsrcDemux *demux;
|
|
guint32 ssrc;
|
|
GstRtpSsrcDemuxPad *dpad;
|
|
GstRTCPPacket packet;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
|
|
|
|
if (!gst_rtcp_buffer_validate (buf))
|
|
goto invalid_rtcp;
|
|
|
|
if (!gst_rtcp_buffer_get_first_packet (buf, &packet))
|
|
goto invalid_rtcp;
|
|
|
|
/* first packet must be SR or RR or else the validate would have failed */
|
|
switch (gst_rtcp_packet_get_type (&packet)) {
|
|
case GST_RTCP_TYPE_SR:
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL,
|
|
NULL);
|
|
break;
|
|
case GST_RTCP_TYPE_RR:
|
|
ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
|
|
break;
|
|
default:
|
|
goto invalid_rtcp;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (demux, "received RTCP of SSRC %08x", ssrc);
|
|
|
|
GST_OBJECT_LOCK (demux);
|
|
dpad = find_demux_pad_for_ssrc (demux, ssrc);
|
|
if (dpad == NULL) {
|
|
GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
|
|
if (!(dpad = create_demux_pad_for_ssrc (demux, ssrc)))
|
|
goto create_failed;
|
|
}
|
|
GST_OBJECT_UNLOCK (demux);
|
|
|
|
/* push to srcpad */
|
|
ret = gst_pad_push (dpad->rtcp_pad, buf);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
invalid_rtcp:
|
|
{
|
|
/* this is fatal and should be filtered earlier */
|
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Dropping invalid RTCP packet"));
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
create_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Could not create new pad"));
|
|
GST_OBJECT_UNLOCK (demux);
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstRtpSsrcDemux *demux;
|
|
gboolean res = FALSE;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
default:
|
|
res = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
gst_object_unref (demux);
|
|
return res;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_ssrc_demux_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstRtpSsrcDemux *demux;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|