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93b433bd16
Original commit message from CVS: * configure.ac: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new), (signal_waiting_threads), (async_jitter_queue_ref), (async_jitter_queue_ref_unlocked), (async_jitter_queue_set_low_threshold), (async_jitter_queue_set_high_threshold), (async_jitter_queue_set_max_queue_length), (async_jitter_queue_get_g_queue), (calculate_ts_diff), (async_jitter_queue_length_ts_units_unlocked), (async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref), (async_jitter_queue_lock), (async_jitter_queue_unlock), (async_jitter_queue_push), (async_jitter_queue_push_unlocked), (async_jitter_queue_push_sorted), (async_jitter_queue_push_sorted_unlocked), (async_jitter_queue_insert_after_unlocked), (async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop), (async_jitter_queue_pop_unlocked), (async_jitter_queue_length), (async_jitter_queue_length_unlocked), (async_jitter_queue_set_flushing_unlocked), (async_jitter_queue_unset_flushing_unlocked), (async_jitter_queue_set_blocking_unlocked): * gst/rtpmanager/async_jitter_queue.h: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (gst_rtp_bin_class_init), (gst_rtp_bin_init), (gst_rtp_bin_finalize), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property), (gst_rtp_bin_change_state), (gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream), (free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init), (gst_rtp_client_class_init), (gst_rtp_client_init), (gst_rtp_client_finalize), (gst_rtp_client_set_property), (gst_rtp_client_get_property), (gst_rtp_client_change_state), (gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad): * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_base_init), (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps), (gst_jitter_buffer_sink_setcaps), (free_func), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_src_activate_push), (gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt), (compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init), (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init), (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain), (gst_rtp_pt_demux_getcaps), (find_pad_for_pt), (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release), (gst_rtp_pt_demux_change_state): * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (gst_rtp_session_change_state), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src), (gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad): * gst/rtpmanager/gstrtpsession.h: Add RTP session management elements. Still in progress.
453 lines
12 KiB
C
453 lines
12 KiB
C
/* GStreamer
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* Copyright (C) <2007> 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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* SECTION:element-rtpsession
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* @short_description: an RTP session manager
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* @see_also: rtpjitterbuffer, rtpbin
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*
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* <refsect2>
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* <para>
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
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* </programlisting>
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-04-02 (0.10.6)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstrtpsession.h"
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/* elementfactory information */
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static const GstElementDetails rtpsession_details =
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GST_ELEMENT_DETAILS ("RTP Session",
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"Filter/Editor/Video",
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"Implement an RTP session",
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"Wim Taymans <wim@fluendo.com>");
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/* sink pads */
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static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
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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 rtpsession_recv_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink",
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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 rtpsession_send_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_sink",
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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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/* src pads */
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static GstStaticPadTemplate rtpsession_recv_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_src",
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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 rtpsession_sync_src_template =
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GST_STATIC_PAD_TEMPLATE ("sync_src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtpsession_send_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
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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 rtpsession_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_src",
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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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/* signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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/* GObject vmethods */
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static void gst_rtp_session_finalize (GObject * object);
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static void gst_rtp_session_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_session_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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/* GstElement vmethods */
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static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
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GstStateChange transition);
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static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name);
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static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
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/*static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 }; */
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GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
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static void
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gst_rtp_session_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 (&rtpsession_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 (&rtpsession_recv_rtcp_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpsession_send_rtp_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 (&rtpsession_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 (&rtpsession_sync_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpsession_rtcp_src_template));
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gst_element_class_set_details (element_class, &rtpsession_details);
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}
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static void
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gst_rtp_session_class_init (GstRTPSessionClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_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_rtp_session_finalize;
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gobject_class->set_property = gst_rtp_session_set_property;
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gobject_class->get_property = gst_rtp_session_get_property;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad);
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gstelement_class->release_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
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}
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static void
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gst_rtp_session_init (GstRTPSession * rtpsession, GstRTPSessionClass * klass)
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{
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}
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static void
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gst_rtp_session_finalize (GObject * object)
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{
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GstRTPSession *rtpsession;
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rtpsession = GST_RTP_SESSION (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_session_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTPSession *rtpsession;
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rtpsession = GST_RTP_SESSION (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_session_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRTPSession *rtpsession;
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rtpsession = GST_RTP_SESSION (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstStateChangeReturn
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gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn res;
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GstRTPSession *rtpsession;
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rtpsession = GST_RTP_SESSION (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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break;
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default:
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break;
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}
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res = parent_class->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return res;
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}
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/* receive a packet from a sender, send it to the RTP session manager and
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* forward the packet on the rtp_src pad
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*/
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static GstFlowReturn
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gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
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{
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GstRTPSession *rtpsession;
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GstFlowReturn ret;
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rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
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/* FIXME, do something */
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ret = gst_pad_push (rtpsession->recv_rtp_src, buffer);
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gst_object_unref (rtpsession);
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return ret;
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}
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/* Receive an RTCP packet from a sender, send it to the RTP session manager and
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* forward the SR packets to the sync_src pad.
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*/
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static GstFlowReturn
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gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer)
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{
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GstRTPSession *rtpsession;
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GstFlowReturn ret;
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rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
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/* FIXME, do something */
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ret = gst_pad_push (rtpsession->sync_src, buffer);
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gst_object_unref (rtpsession);
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return ret;
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}
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/* Recieve an RTP packet to be send to the receivers, send to RTP session
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* manager and forward to send_rtp_src.
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*/
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static GstFlowReturn
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gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
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{
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GstRTPSession *rtpsession;
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GstFlowReturn ret;
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rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
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/* FIXME, do something */
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ret = gst_pad_push (rtpsession->send_rtp_src, buffer);
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gst_object_unref (rtpsession);
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return ret;
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}
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/* Create sinkpad to receive RTP packets from senders. This will also create a
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* srcpad for the RTP packets.
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*/
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static GstPad *
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create_recv_rtp_sink (GstRTPSession * rtpsession)
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{
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rtpsession->recv_rtp_sink =
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gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
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NULL);
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gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
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gst_rtp_session_chain_recv_rtp);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
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rtpsession->recv_rtp_sink);
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rtpsession->recv_rtp_src =
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gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
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NULL);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
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return rtpsession->recv_rtp_sink;
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}
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/* Create a sinkpad to receive RTCP messages from senders, this will also create a
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* sync_src pad for the SR packets.
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*/
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static GstPad *
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create_recv_rtcp_sink (GstRTPSession * rtpsession)
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{
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rtpsession->recv_rtcp_sink =
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gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
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NULL);
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gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
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gst_rtp_session_chain_recv_rtcp);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
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rtpsession->recv_rtcp_sink);
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rtpsession->sync_src =
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gst_pad_new_from_static_template (&rtpsession_sync_src_template, NULL);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
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return rtpsession->recv_rtcp_sink;
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}
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/* Create a sinkpad to receive RTP packets for receivers. This will also create a
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* send_rtp_src pad.
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*/
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static GstPad *
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create_send_rtp_sink (GstRTPSession * rtpsession)
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{
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rtpsession->send_rtp_sink =
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gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
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NULL);
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gst_pad_set_chain_function (rtpsession->send_rtp_sink,
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gst_rtp_session_chain_send_rtp);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
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rtpsession->recv_rtcp_sink);
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rtpsession->send_rtp_src =
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gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
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NULL);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
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return rtpsession->send_rtp_sink;
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}
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/* Create a srcpad with the RTCP packets to send out.
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* This pad will be driven by the RTP session manager when it wants to send out
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* RTCP packets.
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*/
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static GstPad *
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create_rtcp_src (GstRTPSession * rtpsession)
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{
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rtpsession->rtcp_src =
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gst_pad_new_from_static_template (&rtpsession_rtcp_src_template, NULL);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->rtcp_src);
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return rtpsession->rtcp_src;
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}
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static GstPad *
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gst_rtp_session_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstRTPSession *rtpsession;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL);
|
|
|
|
rtpsession = GST_RTP_SESSION (element);
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) {
|
|
if (rtpsession->recv_rtp_sink != NULL)
|
|
goto exists;
|
|
|
|
result = create_recv_rtp_sink (rtpsession);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"recv_rtcp_sink")) {
|
|
if (rtpsession->recv_rtcp_sink != NULL)
|
|
goto exists;
|
|
|
|
result = create_recv_rtcp_sink (rtpsession);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"send_rtp_sink")) {
|
|
if (rtpsession->send_rtp_sink != NULL)
|
|
goto exists;
|
|
|
|
result = create_send_rtp_sink (rtpsession);
|
|
} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src")) {
|
|
if (rtpsession->rtcp_src != NULL)
|
|
goto exists;
|
|
|
|
result = create_rtcp_src (rtpsession);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_warning ("rtpsession: this is not our template");
|
|
return NULL;
|
|
}
|
|
exists:
|
|
{
|
|
g_warning ("rtpsession: pad already requested");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
}
|