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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1325 lines
37 KiB
C
1325 lines
37 KiB
C
/* GStreamer
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* Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
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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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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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/* Element-Checklist-Version: 5 */
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/**
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* SECTION:element-rdtmanager
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* @see_also: GstRtspSrc
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*
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* A simple RTP session manager used internally by rtspsrc.
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*
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* Last reviewed on 2006-06-20 (0.10.4)
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*/
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/* #define HAVE_RTCP */
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#include "gstrdtbuffer.h"
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#include "rdtmanager.h"
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#include "rdtjitterbuffer.h"
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#include <stdio.h>
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GST_DEBUG_CATEGORY_STATIC (rdtmanager_debug);
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#define GST_CAT_DEFAULT (rdtmanager_debug)
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/* GstRDTManager signals and args */
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enum
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{
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SIGNAL_REQUEST_PT_MAP,
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SIGNAL_CLEAR_PT_MAP,
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SIGNAL_ON_NEW_SSRC,
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SIGNAL_ON_SSRC_COLLISION,
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SIGNAL_ON_SSRC_VALIDATED,
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SIGNAL_ON_SSRC_ACTIVE,
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SIGNAL_ON_SSRC_SDES,
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SIGNAL_ON_BYE_SSRC,
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SIGNAL_ON_BYE_TIMEOUT,
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SIGNAL_ON_TIMEOUT,
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LAST_SIGNAL
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};
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#define DEFAULT_LATENCY_MS 200
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enum
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{
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PROP_0,
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PROP_LATENCY
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};
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static GstStaticPadTemplate gst_rdt_manager_recv_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rdt")
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);
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static GstStaticPadTemplate gst_rdt_manager_recv_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate gst_rdt_manager_recv_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rdt")
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);
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static GstStaticPadTemplate gst_rdt_manager_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_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static void gst_rdt_manager_finalize (GObject * object);
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static void gst_rdt_manager_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rdt_manager_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_rdt_manager_query_src (GstPad * pad, GstQuery * query);
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static gboolean gst_rdt_manager_src_activate_push (GstPad * pad,
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gboolean active);
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static GstClock *gst_rdt_manager_provide_clock (GstElement * element);
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static GstStateChangeReturn gst_rdt_manager_change_state (GstElement * element,
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GstStateChange transition);
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static GstPad *gst_rdt_manager_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name);
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static void gst_rdt_manager_release_pad (GstElement * element, GstPad * pad);
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static gboolean gst_rdt_manager_parse_caps (GstRDTManager * rdtmanager,
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GstRDTManagerSession * session, GstCaps * caps);
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static gboolean gst_rdt_manager_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_rdt_manager_chain_rdt (GstPad * pad,
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GstBuffer * buffer);
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static GstFlowReturn gst_rdt_manager_chain_rtcp (GstPad * pad,
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GstBuffer * buffer);
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static void gst_rdt_manager_loop (GstPad * pad);
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static guint gst_rdt_manager_signals[LAST_SIGNAL] = { 0 };
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#define JBUF_LOCK(sess) (g_mutex_lock ((sess)->jbuf_lock))
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#define JBUF_LOCK_CHECK(sess,label) G_STMT_START { \
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JBUF_LOCK (sess); \
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if (sess->srcresult != GST_FLOW_OK) \
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goto label; \
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} G_STMT_END
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#define JBUF_UNLOCK(sess) (g_mutex_unlock ((sess)->jbuf_lock))
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#define JBUF_WAIT(sess) (g_cond_wait ((sess)->jbuf_cond, (sess)->jbuf_lock))
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#define JBUF_WAIT_CHECK(sess,label) G_STMT_START { \
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JBUF_WAIT(sess); \
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if (sess->srcresult != GST_FLOW_OK) \
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goto label; \
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} G_STMT_END
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#define JBUF_SIGNAL(sess) (g_cond_signal ((sess)->jbuf_cond))
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/* Manages the receiving end of the packets.
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*
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* There is one such structure for each RTP session (audio/video/...).
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* We get the RTP/RTCP packets and stuff them into the session manager.
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*/
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struct _GstRDTManagerSession
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{
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/* session id */
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gint id;
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/* the parent bin */
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GstRDTManager *dec;
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gboolean active;
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/* we only support one ssrc and one pt */
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guint32 ssrc;
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guint8 pt;
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gint clock_rate;
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GstCaps *caps;
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gint64 clock_base;
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GstSegment segment;
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/* the last seqnum we pushed out */
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guint32 last_popped_seqnum;
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/* the next expected seqnum */
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guint32 next_seqnum;
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/* last output time */
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GstClockTime last_out_time;
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/* the pads of the session */
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GstPad *recv_rtp_sink;
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GstPad *recv_rtp_src;
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GstPad *recv_rtcp_sink;
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GstPad *rtcp_src;
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GstFlowReturn srcresult;
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gboolean blocked;
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gboolean eos;
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gboolean waiting;
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gboolean discont;
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GstClockID clock_id;
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/* jitterbuffer, lock and cond */
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RDTJitterBuffer *jbuf;
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GMutex *jbuf_lock;
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GCond *jbuf_cond;
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/* some accounting */
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guint64 num_late;
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guint64 num_duplicates;
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};
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/* find a session with the given id */
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static GstRDTManagerSession *
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find_session_by_id (GstRDTManager * rdtmanager, gint id)
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{
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GSList *walk;
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for (walk = rdtmanager->sessions; walk; walk = g_slist_next (walk)) {
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GstRDTManagerSession *sess = (GstRDTManagerSession *) walk->data;
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if (sess->id == id)
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return sess;
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}
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return NULL;
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}
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/* create a session with the given id */
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static GstRDTManagerSession *
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create_session (GstRDTManager * rdtmanager, gint id)
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{
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GstRDTManagerSession *sess;
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sess = g_new0 (GstRDTManagerSession, 1);
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sess->id = id;
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sess->dec = rdtmanager;
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sess->jbuf = rdt_jitter_buffer_new ();
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sess->jbuf_lock = g_mutex_new ();
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sess->jbuf_cond = g_cond_new ();
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rdtmanager->sessions = g_slist_prepend (rdtmanager->sessions, sess);
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return sess;
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}
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static gboolean
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activate_session (GstRDTManager * rdtmanager, GstRDTManagerSession * session,
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guint32 ssrc, guint8 pt)
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{
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GstPadTemplate *templ;
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GstElementClass *klass;
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gchar *name;
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GstCaps *caps;
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GValue ret = { 0 };
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GValue args[3] = { {0}
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, {0}
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, {0}
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};
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GST_DEBUG_OBJECT (rdtmanager, "creating stream");
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session->ssrc = ssrc;
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session->pt = pt;
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/* get pt map */
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g_value_init (&args[0], GST_TYPE_ELEMENT);
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g_value_set_object (&args[0], rdtmanager);
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g_value_init (&args[1], G_TYPE_UINT);
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g_value_set_uint (&args[1], session->id);
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g_value_init (&args[2], G_TYPE_UINT);
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g_value_set_uint (&args[2], pt);
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g_value_init (&ret, GST_TYPE_CAPS);
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g_value_set_boxed (&ret, NULL);
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g_signal_emitv (args, gst_rdt_manager_signals[SIGNAL_REQUEST_PT_MAP], 0,
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&ret);
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g_value_unset (&args[0]);
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g_value_unset (&args[1]);
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g_value_unset (&args[2]);
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caps = (GstCaps *) g_value_dup_boxed (&ret);
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g_value_unset (&ret);
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if (caps)
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gst_rdt_manager_parse_caps (rdtmanager, session, caps);
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name = g_strdup_printf ("recv_rtp_src_%d_%u_%d", session->id, ssrc, pt);
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klass = GST_ELEMENT_GET_CLASS (rdtmanager);
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templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
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session->recv_rtp_src = gst_pad_new_from_template (templ, name);
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g_free (name);
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gst_pad_set_caps (session->recv_rtp_src, caps);
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gst_caps_unref (caps);
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gst_pad_set_element_private (session->recv_rtp_src, session);
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gst_pad_set_query_function (session->recv_rtp_src, gst_rdt_manager_query_src);
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gst_pad_set_activatepush_function (session->recv_rtp_src,
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gst_rdt_manager_src_activate_push);
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gst_pad_set_active (session->recv_rtp_src, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->recv_rtp_src);
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return TRUE;
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}
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static void
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free_session (GstRDTManagerSession * session)
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{
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g_object_unref (session->jbuf);
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g_cond_free (session->jbuf_cond);
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g_mutex_free (session->jbuf_lock);
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g_free (session);
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}
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GST_BOILERPLATE (GstRDTManager, gst_rdt_manager, GstElement, GST_TYPE_ELEMENT);
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static void
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gst_rdt_manager_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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/* sink pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rdt_manager_recv_rtp_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rdt_manager_recv_rtcp_sink_template));
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/* src pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rdt_manager_recv_rtp_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rdt_manager_rtcp_src_template));
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gst_element_class_set_details_simple (element_class, "RTP Decoder",
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"Codec/Parser/Network",
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"Accepts raw RTP and RTCP packets and sends them forward",
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"Wim Taymans <wim@fluendo.com>");
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}
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/* BOXED:UINT,UINT */
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#define g_marshal_value_peek_uint(v) g_value_get_uint (v)
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static void
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gst_rdt_manager_marshal_BOXED__UINT_UINT (GClosure * closure,
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GValue * return_value,
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guint n_param_values,
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const GValue * param_values,
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gpointer invocation_hint, gpointer marshal_data)
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{
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typedef gpointer (*GMarshalFunc_BOXED__UINT_UINT) (gpointer data1,
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guint arg_1, guint arg_2, gpointer data2);
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register GMarshalFunc_BOXED__UINT_UINT callback;
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register GCClosure *cc = (GCClosure *) closure;
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register gpointer data1, data2;
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gpointer v_return;
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g_return_if_fail (return_value != NULL);
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g_return_if_fail (n_param_values == 3);
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if (G_CCLOSURE_SWAP_DATA (closure)) {
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data1 = closure->data;
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data2 = g_value_peek_pointer (param_values + 0);
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} else {
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data1 = g_value_peek_pointer (param_values + 0);
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data2 = closure->data;
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}
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callback =
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(GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data :
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cc->callback);
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v_return = callback (data1,
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g_marshal_value_peek_uint (param_values + 1),
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g_marshal_value_peek_uint (param_values + 2), data2);
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g_value_take_boxed (return_value, v_return);
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}
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static void
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gst_rdt_manager_marshal_VOID__UINT_UINT (GClosure * closure,
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GValue * return_value,
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guint n_param_values,
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const GValue * param_values,
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gpointer invocation_hint, gpointer marshal_data)
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{
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typedef void (*GMarshalFunc_VOID__UINT_UINT) (gpointer data1,
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guint arg_1, guint arg_2, gpointer data2);
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register GMarshalFunc_VOID__UINT_UINT callback;
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register GCClosure *cc = (GCClosure *) closure;
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register gpointer data1, data2;
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g_return_if_fail (n_param_values == 3);
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if (G_CCLOSURE_SWAP_DATA (closure)) {
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data1 = closure->data;
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data2 = g_value_peek_pointer (param_values + 0);
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} else {
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data1 = g_value_peek_pointer (param_values + 0);
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data2 = closure->data;
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}
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callback =
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(GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data :
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cc->callback);
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callback (data1,
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g_marshal_value_peek_uint (param_values + 1),
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g_marshal_value_peek_uint (param_values + 2), data2);
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}
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|
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static void
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gst_rdt_manager_class_init (GstRDTManagerClass * g_class)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRDTManagerClass *klass;
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klass = (GstRDTManagerClass *) g_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gobject_class->finalize = gst_rdt_manager_finalize;
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gobject_class->set_property = gst_rdt_manager_set_property;
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gobject_class->get_property = gst_rdt_manager_get_property;
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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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/**
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* GstRDTManager::request-pt-map:
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* @rdtmanager: the object which received the signal
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* @session: the session
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* @pt: the pt
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*
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* Request the payload type as #GstCaps for @pt in @session.
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*/
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gst_rdt_manager_signals[SIGNAL_REQUEST_PT_MAP] =
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g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, request_pt_map),
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NULL, NULL, gst_rdt_manager_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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/**
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|
* GstRDTManager::clear-pt-map:
|
|
* @rtpbin: the object which received the signal
|
|
*
|
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* Clear all previously cached pt-mapping obtained with
|
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* GstRDTManager::request-pt-map.
|
|
*/
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gst_rdt_manager_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 (GstRDTManagerClass, clear_pt_map),
|
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NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* GstRDTManager::on-bye-ssrc:
|
|
* @rtpbin: the object which received the signal
|
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* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that became inactive because of a BYE packet.
|
|
*/
|
|
gst_rdt_manager_signals[SIGNAL_ON_BYE_SSRC] =
|
|
g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, on_bye_ssrc),
|
|
NULL, NULL, gst_rdt_manager_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRDTManager::on-bye-timeout:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that has timed out because of BYE
|
|
*/
|
|
gst_rdt_manager_signals[SIGNAL_ON_BYE_TIMEOUT] =
|
|
g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, on_bye_timeout),
|
|
NULL, NULL, gst_rdt_manager_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRDTManager::on-timeout:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that has timed out
|
|
*/
|
|
gst_rdt_manager_signals[SIGNAL_ON_TIMEOUT] =
|
|
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRDTManagerClass, on_timeout),
|
|
NULL, NULL, gst_rdt_manager_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
gstelement_class->provide_clock =
|
|
GST_DEBUG_FUNCPTR (gst_rdt_manager_provide_clock);
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_rdt_manager_change_state);
|
|
gstelement_class->request_new_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rdt_manager_request_new_pad);
|
|
gstelement_class->release_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rdt_manager_release_pad);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rdtmanager_debug, "rdtmanager", 0, "RTP decoder");
|
|
}
|
|
|
|
static void
|
|
gst_rdt_manager_init (GstRDTManager * rdtmanager, GstRDTManagerClass * klass)
|
|
{
|
|
rdtmanager->provided_clock = gst_system_clock_obtain ();
|
|
rdtmanager->latency = DEFAULT_LATENCY_MS;
|
|
}
|
|
|
|
static void
|
|
gst_rdt_manager_finalize (GObject * object)
|
|
{
|
|
GstRDTManager *rdtmanager;
|
|
|
|
rdtmanager = GST_RDT_MANAGER (object);
|
|
|
|
g_slist_foreach (rdtmanager->sessions, (GFunc) free_session, NULL);
|
|
g_slist_free (rdtmanager->sessions);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rdt_manager_query_src (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstRDTManager *rdtmanager;
|
|
gboolean res;
|
|
|
|
rdtmanager = GST_RDT_MANAGER (GST_PAD_PARENT (pad));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
GstClockTime latency;
|
|
|
|
latency = rdtmanager->latency * GST_MSECOND;
|
|
|
|
/* we pretend to be live with a 3 second latency */
|
|
gst_query_set_latency (query, TRUE, latency, -1);
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "reporting %" GST_TIME_FORMAT " of latency",
|
|
GST_TIME_ARGS (latency));
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rdt_manager_src_activate_push (GstPad * pad, gboolean active)
|
|
{
|
|
gboolean result = TRUE;
|
|
GstRDTManager *rdtmanager;
|
|
GstRDTManagerSession *session;
|
|
|
|
session = gst_pad_get_element_private (pad);
|
|
rdtmanager = session->dec;
|
|
|
|
if (active) {
|
|
/* allow data processing */
|
|
JBUF_LOCK (session);
|
|
GST_DEBUG_OBJECT (rdtmanager, "Enabling pop on queue");
|
|
/* Mark as non flushing */
|
|
session->srcresult = GST_FLOW_OK;
|
|
gst_segment_init (&session->segment, GST_FORMAT_TIME);
|
|
session->last_popped_seqnum = -1;
|
|
session->last_out_time = -1;
|
|
session->next_seqnum = -1;
|
|
session->eos = FALSE;
|
|
JBUF_UNLOCK (session);
|
|
|
|
/* start pushing out buffers */
|
|
GST_DEBUG_OBJECT (rdtmanager, "Starting task on srcpad");
|
|
gst_pad_start_task (pad, (GstTaskFunction) gst_rdt_manager_loop, pad);
|
|
} else {
|
|
/* make sure all data processing stops ASAP */
|
|
JBUF_LOCK (session);
|
|
/* mark ourselves as flushing */
|
|
session->srcresult = GST_FLOW_WRONG_STATE;
|
|
GST_DEBUG_OBJECT (rdtmanager, "Disabling pop on queue");
|
|
/* this unblocks any waiting pops on the src pad task */
|
|
JBUF_SIGNAL (session);
|
|
/* unlock clock, we just unschedule, the entry will be released by
|
|
* the locking streaming thread. */
|
|
if (session->clock_id)
|
|
gst_clock_id_unschedule (session->clock_id);
|
|
JBUF_UNLOCK (session);
|
|
|
|
/* NOTE this will hardlock if the state change is called from the src pad
|
|
* task thread because we will _join() the thread. */
|
|
GST_DEBUG_OBJECT (rdtmanager, "Stopping task on srcpad");
|
|
result = gst_pad_stop_task (pad);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rdt_manager_handle_data_packet (GstRDTManagerSession * session,
|
|
GstClockTime timestamp, GstRDTPacket * packet)
|
|
{
|
|
GstRDTManager *rdtmanager;
|
|
guint16 seqnum;
|
|
gboolean tail;
|
|
GstFlowReturn res;
|
|
GstBuffer *buffer;
|
|
|
|
rdtmanager = session->dec;
|
|
|
|
res = GST_FLOW_OK;
|
|
|
|
seqnum = 0;
|
|
GST_DEBUG_OBJECT (rdtmanager,
|
|
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
|
|
GST_TIME_ARGS (timestamp));
|
|
|
|
buffer = gst_rdt_packet_to_buffer (packet);
|
|
|
|
JBUF_LOCK_CHECK (session, out_flushing);
|
|
|
|
/* insert the packet into the queue now, FIXME, use seqnum */
|
|
if (!rdt_jitter_buffer_insert (session->jbuf, buffer, timestamp,
|
|
session->clock_rate, &tail))
|
|
goto duplicate;
|
|
|
|
/* signal addition of new buffer when the _loop is waiting. */
|
|
if (session->waiting)
|
|
JBUF_SIGNAL (session);
|
|
|
|
finished:
|
|
JBUF_UNLOCK (session);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
out_flushing:
|
|
{
|
|
res = session->srcresult;
|
|
GST_DEBUG_OBJECT (rdtmanager, "flushing %s", gst_flow_get_name (res));
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
duplicate:
|
|
{
|
|
GST_WARNING_OBJECT (rdtmanager, "Duplicate packet #%d detected, dropping",
|
|
seqnum);
|
|
session->num_duplicates++;
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rdt_manager_parse_caps (GstRDTManager * rdtmanager,
|
|
GstRDTManagerSession * session, GstCaps * caps)
|
|
{
|
|
GstStructure *caps_struct;
|
|
guint val;
|
|
|
|
/* first parse the caps */
|
|
caps_struct = gst_caps_get_structure (caps, 0);
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "got caps");
|
|
|
|
/* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
|
|
* measure the amount of data in the buffer */
|
|
if (!gst_structure_get_int (caps_struct, "clock-rate", &session->clock_rate))
|
|
session->clock_rate = 1000;
|
|
|
|
if (session->clock_rate <= 0)
|
|
goto wrong_rate;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "got clock-rate %d", session->clock_rate);
|
|
|
|
/* gah, clock-base is uint. If we don't have a base, we will use the first
|
|
* buffer timestamp as the base time. This will screw up sync but it's better
|
|
* than nothing. */
|
|
if (gst_structure_get_uint (caps_struct, "clock-base", &val))
|
|
session->clock_base = val;
|
|
else
|
|
session->clock_base = -1;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "got clock-base %" G_GINT64_FORMAT,
|
|
session->clock_base);
|
|
|
|
/* first expected seqnum */
|
|
if (gst_structure_get_uint (caps_struct, "seqnum-base", &val))
|
|
session->next_seqnum = val;
|
|
else
|
|
session->next_seqnum = -1;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "got seqnum-base %d", session->next_seqnum);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
wrong_rate:
|
|
{
|
|
GST_DEBUG_OBJECT (rdtmanager, "Invalid clock-rate %d", session->clock_rate);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rdt_manager_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstRDTManager *rdtmanager;
|
|
GstRDTManagerSession *session;
|
|
gboolean res;
|
|
|
|
rdtmanager = GST_RDT_MANAGER (GST_PAD_PARENT (pad));
|
|
/* find session */
|
|
session = gst_pad_get_element_private (pad);
|
|
|
|
res = gst_rdt_manager_parse_caps (rdtmanager, session, caps);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rdt_manager_chain_rdt (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstFlowReturn res;
|
|
GstRDTManager *rdtmanager;
|
|
GstRDTManagerSession *session;
|
|
GstClockTime timestamp;
|
|
GstRDTPacket packet;
|
|
guint32 ssrc;
|
|
guint8 pt;
|
|
gboolean more;
|
|
|
|
rdtmanager = GST_RDT_MANAGER (GST_PAD_PARENT (pad));
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "got RDT packet");
|
|
|
|
ssrc = 0;
|
|
pt = 0;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "SSRC %08x, PT %d", ssrc, pt);
|
|
|
|
/* find session */
|
|
session = gst_pad_get_element_private (pad);
|
|
|
|
/* see if we have the pad */
|
|
if (!session->active) {
|
|
activate_session (rdtmanager, session, ssrc, pt);
|
|
session->active = TRUE;
|
|
}
|
|
|
|
if (GST_BUFFER_IS_DISCONT (buffer)) {
|
|
GST_DEBUG_OBJECT (rdtmanager, "received discont");
|
|
session->discont = TRUE;
|
|
}
|
|
|
|
res = GST_FLOW_OK;
|
|
|
|
/* take the timestamp of the buffer. This is the time when the packet was
|
|
* received and is used to calculate jitter and clock skew. We will adjust
|
|
* this timestamp with the smoothed value after processing it in the
|
|
* jitterbuffer. */
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
/* bring to running time */
|
|
timestamp = gst_segment_to_running_time (&session->segment, GST_FORMAT_TIME,
|
|
timestamp);
|
|
|
|
more = gst_rdt_buffer_get_first_packet (buffer, &packet);
|
|
while (more) {
|
|
GstRDTType type;
|
|
|
|
type = gst_rdt_packet_get_type (&packet);
|
|
GST_DEBUG_OBJECT (rdtmanager, "Have packet of type %04x", type);
|
|
|
|
if (GST_RDT_IS_DATA_TYPE (type)) {
|
|
GST_DEBUG_OBJECT (rdtmanager, "We have a data packet");
|
|
res = gst_rdt_manager_handle_data_packet (session, timestamp, &packet);
|
|
} else {
|
|
switch (type) {
|
|
default:
|
|
GST_DEBUG_OBJECT (rdtmanager, "Ignoring packet");
|
|
break;
|
|
}
|
|
}
|
|
if (res != GST_FLOW_OK)
|
|
break;
|
|
|
|
more = gst_rdt_packet_move_to_next (&packet);
|
|
}
|
|
|
|
gst_buffer_unref (buffer);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* push packets from the queue to the downstream demuxer */
|
|
static void
|
|
gst_rdt_manager_loop (GstPad * pad)
|
|
{
|
|
GstRDTManager *rdtmanager;
|
|
GstRDTManagerSession *session;
|
|
GstBuffer *buffer;
|
|
GstFlowReturn result;
|
|
|
|
rdtmanager = GST_RDT_MANAGER (GST_PAD_PARENT (pad));
|
|
|
|
session = gst_pad_get_element_private (pad);
|
|
|
|
JBUF_LOCK_CHECK (session, flushing);
|
|
GST_DEBUG_OBJECT (rdtmanager, "Peeking item");
|
|
while (TRUE) {
|
|
/* always wait if we are blocked */
|
|
if (!session->blocked) {
|
|
/* if we have a packet, we can exit the loop and grab it */
|
|
if (rdt_jitter_buffer_num_packets (session->jbuf) > 0)
|
|
break;
|
|
/* no packets but we are EOS, do eos logic */
|
|
if (session->eos)
|
|
goto do_eos;
|
|
}
|
|
/* underrun, wait for packets or flushing now */
|
|
session->waiting = TRUE;
|
|
JBUF_WAIT_CHECK (session, flushing);
|
|
session->waiting = FALSE;
|
|
}
|
|
|
|
buffer = rdt_jitter_buffer_pop (session->jbuf);
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "Got item %p", buffer);
|
|
|
|
if (session->discont) {
|
|
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
|
|
session->discont = FALSE;
|
|
}
|
|
|
|
gst_buffer_set_caps (buffer, GST_PAD_CAPS (session->recv_rtp_src));
|
|
JBUF_UNLOCK (session);
|
|
|
|
result = gst_pad_push (session->recv_rtp_src, buffer);
|
|
if (result != GST_FLOW_OK)
|
|
goto pause;
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (rdtmanager, "we are flushing");
|
|
gst_pad_pause_task (session->recv_rtp_src);
|
|
JBUF_UNLOCK (session);
|
|
return;
|
|
}
|
|
do_eos:
|
|
{
|
|
/* store result, we are flushing now */
|
|
GST_DEBUG_OBJECT (rdtmanager, "We are EOS, pushing EOS downstream");
|
|
session->srcresult = GST_FLOW_UNEXPECTED;
|
|
gst_pad_pause_task (session->recv_rtp_src);
|
|
gst_pad_push_event (session->recv_rtp_src, gst_event_new_eos ());
|
|
JBUF_UNLOCK (session);
|
|
return;
|
|
}
|
|
pause:
|
|
{
|
|
GST_DEBUG_OBJECT (rdtmanager, "pausing task, reason %s",
|
|
gst_flow_get_name (result));
|
|
|
|
JBUF_LOCK (session);
|
|
/* store result */
|
|
session->srcresult = result;
|
|
/* we don't post errors or anything because upstream will do that for us
|
|
* when we pass the return value upstream. */
|
|
gst_pad_pause_task (session->recv_rtp_src);
|
|
JBUF_UNLOCK (session);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rdt_manager_chain_rtcp (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstRDTManager *src;
|
|
|
|
#ifdef HAVE_RTCP
|
|
gboolean valid;
|
|
GstRTCPPacket packet;
|
|
gboolean more;
|
|
#endif
|
|
|
|
src = GST_RDT_MANAGER (GST_PAD_PARENT (pad));
|
|
|
|
GST_DEBUG_OBJECT (src, "got rtcp packet");
|
|
|
|
#ifdef HAVE_RTCP
|
|
valid = gst_rtcp_buffer_validate (buffer);
|
|
if (!valid)
|
|
goto bad_packet;
|
|
|
|
/* position on first packet */
|
|
more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
|
|
while (more) {
|
|
switch (gst_rtcp_packet_get_type (&packet)) {
|
|
case GST_RTCP_TYPE_SR:
|
|
{
|
|
guint32 ssrc, rtptime, packet_count, octet_count;
|
|
guint64 ntptime;
|
|
guint count, i;
|
|
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
|
|
&packet_count, &octet_count);
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
|
|
", RTP %u, PC %u, OC %u", ssrc, ntptime, rtptime, packet_count,
|
|
octet_count);
|
|
|
|
count = gst_rtcp_packet_get_rb_count (&packet);
|
|
for (i = 0; i < count; i++) {
|
|
guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
|
|
guint8 fractionlost;
|
|
gint32 packetslost;
|
|
|
|
gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
|
|
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
|
|
|
|
GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
|
|
", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
|
|
packetslost, exthighestseq, jitter, lsr, dlsr);
|
|
}
|
|
break;
|
|
}
|
|
case GST_RTCP_TYPE_RR:
|
|
{
|
|
guint32 ssrc;
|
|
guint count, i;
|
|
|
|
ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
|
|
|
|
GST_DEBUG_OBJECT (src, "got RR packet: SSRC %08x", ssrc);
|
|
|
|
count = gst_rtcp_packet_get_rb_count (&packet);
|
|
for (i = 0; i < count; i++) {
|
|
guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
|
|
guint8 fractionlost;
|
|
gint32 packetslost;
|
|
|
|
gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
|
|
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
|
|
|
|
GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
|
|
", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
|
|
packetslost, exthighestseq, jitter, lsr, dlsr);
|
|
}
|
|
break;
|
|
}
|
|
case GST_RTCP_TYPE_SDES:
|
|
{
|
|
guint chunks, i, j;
|
|
gboolean more_chunks, more_items;
|
|
|
|
chunks = gst_rtcp_packet_sdes_get_chunk_count (&packet);
|
|
GST_DEBUG_OBJECT (src, "got SDES packet with %d chunks", chunks);
|
|
|
|
more_chunks = gst_rtcp_packet_sdes_first_chunk (&packet);
|
|
i = 0;
|
|
while (more_chunks) {
|
|
guint32 ssrc;
|
|
|
|
ssrc = gst_rtcp_packet_sdes_get_ssrc (&packet);
|
|
|
|
GST_DEBUG_OBJECT (src, "chunk %d, SSRC %08x", i, ssrc);
|
|
|
|
more_items = gst_rtcp_packet_sdes_first_item (&packet);
|
|
j = 0;
|
|
while (more_items) {
|
|
GstRTCPSDESType type;
|
|
guint8 len;
|
|
gchar *data;
|
|
|
|
gst_rtcp_packet_sdes_get_item (&packet, &type, &len, &data);
|
|
|
|
GST_DEBUG_OBJECT (src, "item %d, type %d, len %d, data %s", j,
|
|
type, len, data);
|
|
|
|
more_items = gst_rtcp_packet_sdes_next_item (&packet);
|
|
j++;
|
|
}
|
|
more_chunks = gst_rtcp_packet_sdes_next_chunk (&packet);
|
|
i++;
|
|
}
|
|
break;
|
|
}
|
|
case GST_RTCP_TYPE_BYE:
|
|
{
|
|
guint count, i;
|
|
gchar *reason;
|
|
|
|
reason = gst_rtcp_packet_bye_get_reason (&packet);
|
|
GST_DEBUG_OBJECT (src, "got BYE packet (reason: %s)",
|
|
GST_STR_NULL (reason));
|
|
g_free (reason);
|
|
|
|
count = gst_rtcp_packet_bye_get_ssrc_count (&packet);
|
|
for (i = 0; i < count; i++) {
|
|
guint32 ssrc;
|
|
|
|
|
|
ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, i);
|
|
|
|
GST_DEBUG_OBJECT (src, "SSRC: %08x", ssrc);
|
|
}
|
|
break;
|
|
}
|
|
case GST_RTCP_TYPE_APP:
|
|
GST_DEBUG_OBJECT (src, "got APP packet");
|
|
break;
|
|
default:
|
|
GST_WARNING_OBJECT (src, "got unknown RTCP packet");
|
|
break;
|
|
}
|
|
more = gst_rtcp_packet_move_to_next (&packet);
|
|
}
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
|
|
bad_packet:
|
|
{
|
|
GST_WARNING_OBJECT (src, "got invalid RTCP packet");
|
|
return GST_FLOW_OK;
|
|
}
|
|
#else
|
|
return GST_FLOW_OK;
|
|
#endif
|
|
}
|
|
|
|
static void
|
|
gst_rdt_manager_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRDTManager *src;
|
|
|
|
src = GST_RDT_MANAGER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
src->latency = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rdt_manager_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstRDTManager *src;
|
|
|
|
src = GST_RDT_MANAGER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
g_value_set_uint (value, src->latency);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstClock *
|
|
gst_rdt_manager_provide_clock (GstElement * element)
|
|
{
|
|
GstRDTManager *rdtmanager;
|
|
|
|
rdtmanager = GST_RDT_MANAGER (element);
|
|
|
|
return GST_CLOCK_CAST (gst_object_ref (rdtmanager->provided_clock));
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rdt_manager_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* we're NO_PREROLL when going to PAUSED */
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* Create a pad for receiving RTP for the session in @name
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtp (GstRDTManager * rdtmanager, GstPadTemplate * templ,
|
|
const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRDTManagerSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "finding session %d", sessid);
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rdtmanager, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rdtmanager, "creating session %d", sessid);
|
|
/* create session now */
|
|
session = create_session (rdtmanager, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
/* check if pad was requested */
|
|
if (session->recv_rtp_sink != NULL)
|
|
goto existed;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "getting RTP sink pad");
|
|
|
|
session->recv_rtp_sink = gst_pad_new_from_template (templ, name);
|
|
gst_pad_set_element_private (session->recv_rtp_sink, session);
|
|
gst_pad_set_setcaps_function (session->recv_rtp_sink,
|
|
gst_rdt_manager_setcaps);
|
|
gst_pad_set_chain_function (session->recv_rtp_sink,
|
|
gst_rdt_manager_chain_rdt);
|
|
gst_pad_set_active (session->recv_rtp_sink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->recv_rtp_sink);
|
|
|
|
return session->recv_rtp_sink;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rdtmanager: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rdtmanager: recv_rtp pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for receiving RTCP for the session in @name
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtcp (GstRDTManager * rdtmanager, GstPadTemplate * templ,
|
|
const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRDTManagerSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "finding session %d", sessid);
|
|
|
|
/* get the session, it must exist or we error */
|
|
session = find_session_by_id (rdtmanager, sessid);
|
|
if (!session)
|
|
goto no_session;
|
|
|
|
/* check if pad was requested */
|
|
if (session->recv_rtcp_sink != NULL)
|
|
goto existed;
|
|
|
|
GST_DEBUG_OBJECT (rdtmanager, "getting RTCP sink pad");
|
|
|
|
session->recv_rtcp_sink = gst_pad_new_from_template (templ, name);
|
|
gst_pad_set_element_private (session->recv_rtp_sink, session);
|
|
gst_pad_set_chain_function (session->recv_rtcp_sink,
|
|
gst_rdt_manager_chain_rtcp);
|
|
gst_pad_set_active (session->recv_rtcp_sink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->recv_rtcp_sink);
|
|
|
|
return session->recv_rtcp_sink;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rdtmanager: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("rdtmanager: no session with id %d", sessid);
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rdtmanager: recv_rtcp pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for sending RTCP for the session in @name
|
|
*/
|
|
static GstPad *
|
|
create_rtcp (GstRDTManager * rdtmanager, GstPadTemplate * templ,
|
|
const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRDTManagerSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "rtcp_src_%d", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rdtmanager, sessid);
|
|
if (!session)
|
|
goto no_session;
|
|
|
|
/* check if pad was requested */
|
|
if (session->rtcp_src != NULL)
|
|
goto existed;
|
|
|
|
session->rtcp_src = gst_pad_new_from_template (templ, name);
|
|
gst_pad_set_active (session->rtcp_src, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rdtmanager), session->rtcp_src);
|
|
|
|
return session->rtcp_src;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rdtmanager: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("rdtmanager: session with id %d does not exist", sessid);
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rdtmanager: rtcp_src pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/*
|
|
*/
|
|
static GstPad *
|
|
gst_rdt_manager_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstRDTManager *rdtmanager;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RDT_MANAGER (element), NULL);
|
|
|
|
rdtmanager = GST_RDT_MANAGER (element);
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
|
|
result = create_recv_rtp (rdtmanager, templ, name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"recv_rtcp_sink_%d")) {
|
|
result = create_recv_rtcp (rdtmanager, templ, name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%d")) {
|
|
result = create_rtcp (rdtmanager, templ, name);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_warning ("rdtmanager: this is not our template");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rdt_manager_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
}
|
|
|
|
gboolean
|
|
gst_rdt_manager_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rdtmanager",
|
|
GST_RANK_NONE, GST_TYPE_RDT_MANAGER);
|
|
}
|