Add RTP session management elements. Still in progress.

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.

1 files changed, 453 insertions, 0 deletions

diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
new file mode 100644
index 00000000..47df756f
--- /dev/null
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -0,0 +1,453 @@
+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-rtpsession
+ * @short_description: an RTP session manager
+ * @see_also: rtpjitterbuffer, rtpbin
+ *
+ * <refsect2>
+ * <para>
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ *
+ * Last reviewed on 2007-04-02 (0.10.6)
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include "gstrtpsession.h"
+
+/* elementfactory information */
+static const GstElementDetails rtpsession_details =
+GST_ELEMENT_DETAILS ("RTP Session",
+    "Filter/Editor/Video",
+    "Implement an RTP session",
+    "Wim Taymans <wim@fluendo.com>");
+
+/* sink pads */
+static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
+GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template =
+GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtcp")
+    );
+
+static GstStaticPadTemplate rtpsession_send_rtp_sink_template =
+GST_STATIC_PAD_TEMPLATE ("send_rtp_sink",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+/* src pads */
+static GstStaticPadTemplate rtpsession_recv_rtp_src_template =
+GST_STATIC_PAD_TEMPLATE ("recv_rtp_src",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+static GstStaticPadTemplate rtpsession_sync_src_template =
+GST_STATIC_PAD_TEMPLATE ("sync_src",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("application/x-rtcp")
+    );
+
+static GstStaticPadTemplate rtpsession_send_rtp_src_template =
+GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+static GstStaticPadTemplate rtpsession_rtcp_src_template =
+GST_STATIC_PAD_TEMPLATE ("rtcp_src",
+    GST_PAD_SRC,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtcp")
+    );
+
+/* signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0
+};
+
+/* GObject vmethods */
+static void gst_rtp_session_finalize (GObject * object);
+static void gst_rtp_session_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtp_session_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+/* GstElement vmethods */
+static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
+    GstStateChange transition);
+static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
+    GstPadTemplate * templ, const gchar * name);
+static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
+
+/*static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 }; */
+
+GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
+
+static void
+gst_rtp_session_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  /* sink pads */
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
+
+  /* src pads */
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpsession_sync_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpsession_rtcp_src_template));
+
+  gst_element_class_set_details (element_class, &rtpsession_details);
+}
+
+static void
+gst_rtp_session_class_init (GstRTPSessionClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  gobject_class->finalize = gst_rtp_session_finalize;
+  gobject_class->set_property = gst_rtp_session_set_property;
+  gobject_class->get_property = gst_rtp_session_get_property;
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
+  gstelement_class->request_new_pad =
+      GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad);
+  gstelement_class->release_pad =
+      GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
+}
+
+static void
+gst_rtp_session_init (GstRTPSession * rtpsession, GstRTPSessionClass * klass)
+{
+}
+
+static void
+gst_rtp_session_finalize (GObject * object)
+{
+  GstRTPSession *rtpsession;
+
+  rtpsession = GST_RTP_SESSION (object);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_rtp_session_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTPSession *rtpsession;
+
+  rtpsession = GST_RTP_SESSION (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_session_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTPSession *rtpsession;
+
+  rtpsession = GST_RTP_SESSION (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn res;
+  GstRTPSession *rtpsession;
+
+  rtpsession = GST_RTP_SESSION (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      break;
+    default:
+      break;
+  }
+
+  res = parent_class->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+  return res;
+}
+
+/* receive a packet from a sender, send it to the RTP session manager and
+ * forward the packet on the rtp_src pad
+ */
+static GstFlowReturn
+gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
+{
+  GstRTPSession *rtpsession;
+  GstFlowReturn ret;
+
+  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
+
+  /* FIXME, do something */
+  ret = gst_pad_push (rtpsession->recv_rtp_src, buffer);
+
+  gst_object_unref (rtpsession);
+
+  return ret;
+}
+
+/* Receive an RTCP packet from a sender, send it to the RTP session manager and
+ * forward the SR packets to the sync_src pad.
+ */
+static GstFlowReturn
+gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer)
+{
+  GstRTPSession *rtpsession;
+  GstFlowReturn ret;
+
+  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
+
+  /* FIXME, do something */
+  ret = gst_pad_push (rtpsession->sync_src, buffer);
+
+  gst_object_unref (rtpsession);
+
+  return ret;
+}
+
+/* Recieve an RTP packet to be send to the receivers, send to RTP session
+ * manager and forward to send_rtp_src.
+ */
+static GstFlowReturn
+gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
+{
+  GstRTPSession *rtpsession;
+  GstFlowReturn ret;
+
+  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
+
+  /* FIXME, do something */
+  ret = gst_pad_push (rtpsession->send_rtp_src, buffer);
+
+  gst_object_unref (rtpsession);
+
+  return ret;
+}
+
+
+/* Create sinkpad to receive RTP packets from senders. This will also create a
+ * srcpad for the RTP packets.
+ */
+static GstPad *
+create_recv_rtp_sink (GstRTPSession * rtpsession)
+{
+  rtpsession->recv_rtp_sink =
+      gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
+      NULL);
+  gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
+      gst_rtp_session_chain_recv_rtp);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
+      rtpsession->recv_rtp_sink);
+
+  rtpsession->recv_rtp_src =
+      gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
+      NULL);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
+
+  return rtpsession->recv_rtp_sink;
+}
+
+/* Create a sinkpad to receive RTCP messages from senders, this will also create a
+ * sync_src pad for the SR packets.
+ */
+static GstPad *
+create_recv_rtcp_sink (GstRTPSession * rtpsession)
+{
+  rtpsession->recv_rtcp_sink =
+      gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
+      NULL);
+  gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
+      gst_rtp_session_chain_recv_rtcp);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
+      rtpsession->recv_rtcp_sink);
+
+  rtpsession->sync_src =
+      gst_pad_new_from_static_template (&rtpsession_sync_src_template, NULL);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
+
+  return rtpsession->recv_rtcp_sink;
+}
+
+/* Create a sinkpad to receive RTP packets for receivers. This will also create a
+ * send_rtp_src pad.
+ */
+static GstPad *
+create_send_rtp_sink (GstRTPSession * rtpsession)
+{
+  rtpsession->send_rtp_sink =
+      gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
+      NULL);
+  gst_pad_set_chain_function (rtpsession->send_rtp_sink,
+      gst_rtp_session_chain_send_rtp);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
+      rtpsession->recv_rtcp_sink);
+
+  rtpsession->send_rtp_src =
+      gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
+      NULL);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
+
+  return rtpsession->send_rtp_sink;
+}
+
+/* Create a srcpad with the RTCP packets to send out.
+ * This pad will be driven by the RTP session manager when it wants to send out
+ * RTCP packets.
+ */
+static GstPad *
+create_rtcp_src (GstRTPSession * rtpsession)
+{
+  rtpsession->rtcp_src =
+      gst_pad_new_from_static_template (&rtpsession_rtcp_src_template, NULL);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->rtcp_src);
+
+  return rtpsession->rtcp_src;
+}
+
+static GstPad *
+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)
+{
+}