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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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895 lines
25 KiB
C
895 lines
25 KiB
C
/* GStreamer
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* Copyright (C) <2005,2006> Wim Taymans <wim.taymans@gmail.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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-rtpdec
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* @title: rtpdec
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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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/* #define HAVE_RTCP */
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#include <gst/rtp/gstrtpbuffer.h>
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#ifdef HAVE_RTCP
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#include <gst/rtp/gstrtcpbuffer.h>
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#endif
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#include "gstrtspelements.h"
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#include "gstrtpdec.h"
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#include <stdio.h>
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GST_DEBUG_CATEGORY_STATIC (rtpdec_debug);
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#define GST_CAT_DEFAULT (rtpdec_debug)
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/* GstRTPDec 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_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_rtp_dec_recv_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate gst_rtp_dec_recv_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
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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_rtp_dec_recv_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
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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 gst_rtp_dec_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_src_%u",
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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_rtp_dec_finalize (GObject * object);
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static void gst_rtp_dec_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rtp_dec_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstClock *gst_rtp_dec_provide_clock (GstElement * element);
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static GstStateChangeReturn gst_rtp_dec_change_state (GstElement * element,
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GstStateChange transition);
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static GstPad *gst_rtp_dec_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
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static void gst_rtp_dec_release_pad (GstElement * element, GstPad * pad);
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static GstFlowReturn gst_rtp_dec_chain_rtp (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static GstFlowReturn gst_rtp_dec_chain_rtcp (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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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 _GstRTPDecSession
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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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GstRTPDec *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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GstCaps *caps;
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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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};
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/* find a session with the given id */
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static GstRTPDecSession *
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find_session_by_id (GstRTPDec * rtpdec, gint id)
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{
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GSList *walk;
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for (walk = rtpdec->sessions; walk; walk = g_slist_next (walk)) {
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GstRTPDecSession *sess = (GstRTPDecSession *) 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 GstRTPDecSession *
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create_session (GstRTPDec * rtpdec, gint id)
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{
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GstRTPDecSession *sess;
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sess = g_new0 (GstRTPDecSession, 1);
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sess->id = id;
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sess->dec = rtpdec;
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rtpdec->sessions = g_slist_prepend (rtpdec->sessions, sess);
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return sess;
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}
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static void
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free_session (GstRTPDecSession * session)
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{
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g_free (session);
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}
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static guint gst_rtp_dec_signals[LAST_SIGNAL] = { 0 };
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#define gst_rtp_dec_parent_class parent_class
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G_DEFINE_TYPE (GstRTPDec, gst_rtp_dec, GST_TYPE_ELEMENT);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpdec, "rtpdec", GST_RANK_NONE,
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GST_TYPE_RTP_DEC, rtsp_element_init (plugin));
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static void
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gst_rtp_dec_class_init (GstRTPDecClass * g_class)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPDecClass *klass;
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klass = (GstRTPDecClass *) g_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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GST_DEBUG_CATEGORY_INIT (rtpdec_debug, "rtpdec", 0, "RTP decoder");
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gobject_class->finalize = gst_rtp_dec_finalize;
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gobject_class->set_property = gst_rtp_dec_set_property;
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gobject_class->get_property = gst_rtp_dec_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 | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTPDec::request-pt-map:
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* @rtpdec: 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_rtp_dec_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 (GstRTPDecClass, request_pt_map), NULL,
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NULL, NULL, GST_TYPE_CAPS, 2, G_TYPE_UINT, G_TYPE_UINT);
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gst_rtp_dec_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 (GstRTPDecClass, clear_pt_map), NULL,
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NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
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/**
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* GstRTPDec::on-new-ssrc:
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* @rtpbin: the object which received the signal
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* @session: the session
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* @ssrc: the SSRC
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*
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* Notify of a new SSRC that entered @session.
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*/
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gst_rtp_dec_signals[SIGNAL_ON_NEW_SSRC] =
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g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_new_ssrc), NULL,
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NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
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/**
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* GstRTPDec::on-ssrc_collision:
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* @rtpbin: the object which received the signal
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* @session: the session
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* @ssrc: the SSRC
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*
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* Notify when we have an SSRC collision
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*/
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gst_rtp_dec_signals[SIGNAL_ON_SSRC_COLLISION] =
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g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_collision),
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NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
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/**
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* GstRTPDec::on-ssrc_validated:
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* @rtpbin: the object which received the signal
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* @session: the session
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* @ssrc: the SSRC
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*
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* Notify of a new SSRC that became validated.
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*/
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gst_rtp_dec_signals[SIGNAL_ON_SSRC_VALIDATED] =
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g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_validated),
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NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
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/**
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* GstRTPDec::on-bye-ssrc:
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* @rtpbin: the object which received the signal
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* @session: the session
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* @ssrc: the SSRC
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*
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* Notify of an SSRC that became inactive because of a BYE packet.
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*/
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gst_rtp_dec_signals[SIGNAL_ON_BYE_SSRC] =
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g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_ssrc), NULL,
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NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
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/**
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* GstRTPDec::on-bye-timeout:
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* @rtpbin: the object which received the signal
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* @session: the session
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* @ssrc: the SSRC
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*
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* Notify of an SSRC that has timed out because of BYE
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*/
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gst_rtp_dec_signals[SIGNAL_ON_BYE_TIMEOUT] =
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g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_timeout),
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NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
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/**
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* GstRTPDec::on-timeout:
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* @rtpbin: the object which received the signal
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* @session: the session
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* @ssrc: the SSRC
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*
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* Notify of an SSRC that has timed out
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*/
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gst_rtp_dec_signals[SIGNAL_ON_TIMEOUT] =
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g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_timeout), NULL,
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NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
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gstelement_class->provide_clock =
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GST_DEBUG_FUNCPTR (gst_rtp_dec_provide_clock);
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_dec_change_state);
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_dec_request_new_pad);
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gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_dec_release_pad);
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/* sink pads */
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_dec_recv_rtp_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_dec_recv_rtcp_sink_template);
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/* src pads */
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_dec_recv_rtp_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_dec_rtcp_src_template);
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gst_element_class_set_static_metadata (gstelement_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.taymans@gmail.com>");
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}
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static void
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gst_rtp_dec_init (GstRTPDec * rtpdec)
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{
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rtpdec->provided_clock = gst_system_clock_obtain ();
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rtpdec->latency = DEFAULT_LATENCY_MS;
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GST_OBJECT_FLAG_SET (rtpdec, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
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}
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static void
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gst_rtp_dec_finalize (GObject * object)
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{
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GstRTPDec *rtpdec;
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rtpdec = GST_RTP_DEC (object);
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gst_object_unref (rtpdec->provided_clock);
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g_slist_foreach (rtpdec->sessions, (GFunc) free_session, NULL);
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g_slist_free (rtpdec->sessions);
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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_dec_query_src (GstPad * pad, GstObject * parent, GstQuery * query)
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{
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gboolean res;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_LATENCY:
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{
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/* we pretend to be live with a 3 second latency */
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/* FIXME: Do we really have infinite maximum latency? */
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gst_query_set_latency (query, TRUE, 3 * GST_SECOND, -1);
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res = TRUE;
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break;
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}
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default:
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res = gst_pad_query_default (pad, parent, query);
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break;
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}
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return res;
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}
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static GstFlowReturn
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gst_rtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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{
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GstFlowReturn res;
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GstRTPDec *rtpdec;
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GstRTPDecSession *session;
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guint32 ssrc;
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guint8 pt;
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GstRTPBuffer rtp = { NULL, };
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rtpdec = GST_RTP_DEC (parent);
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GST_DEBUG_OBJECT (rtpdec, "got rtp packet");
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if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
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goto bad_packet;
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ssrc = gst_rtp_buffer_get_ssrc (&rtp);
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pt = gst_rtp_buffer_get_payload_type (&rtp);
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gst_rtp_buffer_unmap (&rtp);
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GST_DEBUG_OBJECT (rtpdec, "SSRC %08x, PT %d", ssrc, pt);
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/* find session */
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session = gst_pad_get_element_private (pad);
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|
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/* see if we have the pad */
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if (!session->active) {
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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 (rtpdec, "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], rtpdec);
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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_rtp_dec_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
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caps = (GstCaps *) g_value_get_boxed (&ret);
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|
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name = g_strdup_printf ("recv_rtp_src_%u_%u_%u", session->id, ssrc, pt);
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klass = GST_ELEMENT_GET_CLASS (rtpdec);
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templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
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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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|
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gst_pad_set_caps (session->recv_rtp_src, caps);
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|
|
gst_pad_set_element_private (session->recv_rtp_src, session);
|
|
gst_pad_set_query_function (session->recv_rtp_src, gst_rtp_dec_query_src);
|
|
gst_pad_set_active (session->recv_rtp_src, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtp_src);
|
|
|
|
session->active = TRUE;
|
|
}
|
|
|
|
res = gst_pad_push (session->recv_rtp_src, buffer);
|
|
|
|
return res;
|
|
|
|
bad_packet:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpdec, STREAM, DECODE, (NULL),
|
|
("RTP packet did not validate, dropping"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer)
|
|
{
|
|
GstRTPDec *src;
|
|
|
|
#ifdef HAVE_RTCP
|
|
gboolean valid;
|
|
GstRTCPPacket packet;
|
|
gboolean more;
|
|
#endif
|
|
|
|
src = GST_RTP_DEC (parent);
|
|
|
|
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");
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
#else
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
#endif
|
|
}
|
|
|
|
static void
|
|
gst_rtp_dec_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTPDec *src;
|
|
|
|
src = GST_RTP_DEC (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_rtp_dec_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstRTPDec *src;
|
|
|
|
src = GST_RTP_DEC (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_rtp_dec_provide_clock (GstElement * element)
|
|
{
|
|
GstRTPDec *rtpdec;
|
|
|
|
rtpdec = GST_RTP_DEC (element);
|
|
|
|
return GST_CLOCK_CAST (gst_object_ref (rtpdec->provided_clock));
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_dec_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 (GstRTPDec * rtpdec, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRTPDecSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rtpdec, "finding session %d", sessid);
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpdec, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpdec, "creating session %d", sessid);
|
|
/* create session now */
|
|
session = create_session (rtpdec, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
/* check if pad was requested */
|
|
if (session->recv_rtp_sink != NULL)
|
|
goto existed;
|
|
|
|
GST_DEBUG_OBJECT (rtpdec, "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_chain_function (session->recv_rtp_sink, gst_rtp_dec_chain_rtp);
|
|
gst_pad_set_active (session->recv_rtp_sink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtp_sink);
|
|
|
|
return session->recv_rtp_sink;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpdec: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rtpdec: 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 (GstRTPDec * rtpdec, GstPadTemplate * templ,
|
|
const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRTPDecSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rtpdec, "finding session %d", sessid);
|
|
|
|
/* get the session, it must exist or we error */
|
|
session = find_session_by_id (rtpdec, sessid);
|
|
if (!session)
|
|
goto no_session;
|
|
|
|
/* check if pad was requested */
|
|
if (session->recv_rtcp_sink != NULL)
|
|
goto existed;
|
|
|
|
GST_DEBUG_OBJECT (rtpdec, "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_rtp_dec_chain_rtcp);
|
|
gst_pad_set_active (session->recv_rtcp_sink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtcp_sink);
|
|
|
|
return session->recv_rtcp_sink;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpdec: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("rtpdec: no session with id %d", sessid);
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rtpdec: 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 (GstRTPDec * rtpdec, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRTPDecSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "rtcp_src_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpdec, 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 (rtpdec), session->rtcp_src);
|
|
|
|
return session->rtcp_src;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpdec: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("rtpdec: session with id %d does not exist", sessid);
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rtpdec: rtcp_src pad already requested for session %d", sessid);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/*
|
|
*/
|
|
static GstPad *
|
|
gst_rtp_dec_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
|
|
{
|
|
GstRTPDec *rtpdec;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_DEC (element), NULL);
|
|
|
|
rtpdec = GST_RTP_DEC (element);
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
|
|
result = create_recv_rtp (rtpdec, templ, name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"recv_rtcp_sink_%u")) {
|
|
result = create_recv_rtcp (rtpdec, templ, name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%u")) {
|
|
result = create_rtcp (rtpdec, templ, name);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_warning ("rtpdec: this is not our template");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_dec_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
}
|