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, 482 insertions, 0 deletions

diff --git a/gst/rtpmanager/gstrtpclient.c b/gst/rtpmanager/gstrtpclient.c
new file mode 100644
index 00000000..2984d3f9
--- /dev/null
+++ b/gst/rtpmanager/gstrtpclient.c
@@ -0,0 +1,482 @@
+/* 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-rtpclient
+ * @short_description: handle media from one RTP client
+ * @see_also: rtpjitterbuffer, rtpbin, rtpsession
+ *
+ * <refsect2>
+ * <para>
+ * This element handles RTP data from one client. It accepts multiple RTP streams that
+ * should be synchronized together.
+ * </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 <string.h>
+
+#include "gstrtpclient.h"
+
+/* elementfactory information */
+static const GstElementDetails rtpclient_details =
+GST_ELEMENT_DETAILS ("RTP Client",
+    "Filter/Editor/Video",
+    "Implement an RTP client",
+    "Wim Taymans <wim@fluendo.com>");
+
+/* sink pads */
+static GstStaticPadTemplate rtpclient_rtp_sink_template =
+GST_STATIC_PAD_TEMPLATE ("rtp_sink_%d",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+static GstStaticPadTemplate rtpclient_sync_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sync_sink_%d",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtcp")
+    );
+
+/* src pads */
+static GstStaticPadTemplate rtpclient_rtp_src_template =
+GST_STATIC_PAD_TEMPLATE ("rtp_src_%d_%d",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+#define GST_RTP_CLIENT_GET_PRIVATE(obj)  \
+   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_CLIENT, GstRTPClientPrivate))
+
+struct _GstRTPClientPrivate
+{
+};
+
+/* all the info needed to handle the stream with SSRC */
+typedef struct
+{
+  GstRTPClient *client;
+
+  /* the SSRC of this stream */
+  guint32 ssrc;
+
+  /* RTP and RTCP in */
+  GstPad *rtp_sink;
+  GstPad *sync_sink;
+
+  /* the jitterbuffer */
+  GstElement *jitterbuffer;
+  /* the payload demuxer */
+  GstElement *ptdemux;
+  /* the new-pad signal */
+  gulong new_pad_sig;
+} GstRTPClientStream;
+
+/* the PT demuxer found a new payload type */
+static void
+new_pad (GstElement * element, GstPad * pad, GstRTPClientStream * stream)
+{
+}
+
+/* create a new stream for SSRC.
+ *
+ * We create a jitterbuffer and an payload demuxer for the SSRC. The sinkpad of
+ * the jitterbuffer is ghosted to the bin. We connect a pad-added signal to
+ * rtpptdemux so that we can ghost the payload pads outside.
+ *
+ *       +-----------------+     +---------------+
+ *       | rtpjitterbuffer |     |  rtpptdemux   |
+ *   +- sink              src - sink             |
+ *  /    +-----------------+     +---------------+
+ *
+ */
+static GstRTPClientStream *
+create_stream (GstRTPClient * rtpclient, guint32 ssrc)
+{
+  GstRTPClientStream *stream;
+  gchar *name;
+  GstPad *srcpad, *sinkpad;
+  GstPadLinkReturn res;
+
+  stream = g_new0 (GstRTPClientStream, 1);
+  stream->ssrc = ssrc;
+  stream->client = rtpclient;
+
+  stream->jitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL);
+  if (!stream->jitterbuffer)
+    goto no_jitterbuffer;
+
+  stream->ptdemux = gst_element_factory_make ("rtpptdemux", NULL);
+  if (!stream->ptdemux)
+    goto no_ptdemux;
+
+  /* add elements to bin */
+  gst_bin_add (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
+  gst_bin_add (GST_BIN_CAST (rtpclient), stream->ptdemux);
+
+  /* link jitterbuffer and PT demuxer */
+  srcpad = gst_element_get_pad (stream->jitterbuffer, "src");
+  sinkpad = gst_element_get_pad (stream->ptdemux, "sink");
+  res = gst_pad_link (srcpad, sinkpad);
+  gst_object_unref (srcpad);
+  gst_object_unref (sinkpad);
+
+  if (res != GST_PAD_LINK_OK)
+    goto could_not_link;
+
+  /* add stream to list */
+  rtpclient->streams = g_list_prepend (rtpclient->streams, stream);
+
+  /* ghost sinkpad */
+  name = g_strdup_printf ("rtp_sink_%d", ssrc);
+  sinkpad = gst_element_get_pad (stream->jitterbuffer, "sink");
+  stream->rtp_sink = gst_ghost_pad_new (name, sinkpad);
+  gst_object_unref (sinkpad);
+  g_free (name);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->rtp_sink);
+
+  /* add signal to ptdemuxer */
+  stream->new_pad_sig =
+      g_signal_connect (G_OBJECT (stream->ptdemux), "pad-added",
+      G_CALLBACK (new_pad), stream);
+
+  return stream;
+
+  /* ERRORS */
+no_jitterbuffer:
+  {
+    g_free (stream);
+    g_warning ("could not create rtpjitterbuffer element");
+    return NULL;
+  }
+no_ptdemux:
+  {
+    gst_object_unref (stream->jitterbuffer);
+    g_free (stream);
+    g_warning ("could not create rtpptdemux element");
+    return NULL;
+  }
+could_not_link:
+  {
+    gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
+    gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux);
+    g_free (stream);
+    g_warning ("could not link jitterbuffer and rtpptdemux element");
+    return NULL;
+  }
+}
+
+#if 0
+static void
+free_stream (GstRTPClientStream * stream)
+{
+  gst_object_unref (stream->jitterbuffer);
+  g_free (stream);
+}
+#endif
+
+/* find the stream for the given SSRC, return NULL if the stream did not exist
+ */
+static GstRTPClientStream *
+find_stream_by_ssrc (GstRTPClient * client, guint32 ssrc)
+{
+  GstRTPClientStream *stream;
+  GList *walk;
+
+  for (walk = client->streams; walk; walk = g_list_next (walk)) {
+    stream = (GstRTPClientStream *) walk->data;
+    if (stream->ssrc == ssrc)
+      return stream;
+  }
+  return NULL;
+}
+
+/* signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0
+};
+
+/* GObject vmethods */
+static void gst_rtp_client_finalize (GObject * object);
+static void gst_rtp_client_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtp_client_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+/* GstElement vmethods */
+static GstStateChangeReturn gst_rtp_client_change_state (GstElement * element,
+    GstStateChange transition);
+static GstPad *gst_rtp_client_request_new_pad (GstElement * element,
+    GstPadTemplate * templ, const gchar * name);
+static void gst_rtp_client_release_pad (GstElement * element, GstPad * pad);
+
+/*static guint gst_rtp_client_signals[LAST_SIGNAL] = { 0 }; */
+
+GST_BOILERPLATE (GstRTPClient, gst_rtp_client, GstBin, GST_TYPE_BIN);
+
+static void
+gst_rtp_client_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 (&rtpclient_rtp_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpclient_sync_sink_template));
+
+  /* src pads */
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&rtpclient_rtp_src_template));
+
+  gst_element_class_set_details (element_class, &rtpclient_details);
+}
+
+static void
+gst_rtp_client_class_init (GstRTPClientClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  g_type_class_add_private (klass, sizeof (GstRTPClientPrivate));
+
+  gobject_class->finalize = gst_rtp_client_finalize;
+  gobject_class->set_property = gst_rtp_client_set_property;
+  gobject_class->get_property = gst_rtp_client_get_property;
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_rtp_client_change_state);
+  gstelement_class->request_new_pad =
+      GST_DEBUG_FUNCPTR (gst_rtp_client_request_new_pad);
+  gstelement_class->release_pad =
+      GST_DEBUG_FUNCPTR (gst_rtp_client_release_pad);
+}
+
+static void
+gst_rtp_client_init (GstRTPClient * rtpclient, GstRTPClientClass * klass)
+{
+  rtpclient->priv = GST_RTP_CLIENT_GET_PRIVATE (rtpclient);
+}
+
+static void
+gst_rtp_client_finalize (GObject * object)
+{
+  GstRTPClient *rtpclient;
+
+  rtpclient = GST_RTP_CLIENT (object);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_rtp_client_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTPClient *rtpclient;
+
+  rtpclient = GST_RTP_CLIENT (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_client_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTPClient *rtpclient;
+
+  rtpclient = GST_RTP_CLIENT (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_rtp_client_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn res;
+  GstRTPClient *rtpclient;
+
+  rtpclient = GST_RTP_CLIENT (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 = GST_ELEMENT_CLASS (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;
+}
+
+/* We have 2 request pads (rtp_sink_%d and sync_sink_%d), the %d is assumed to
+ * be the SSRC of the stream.
+ *
+ * We require that the rtp pad is requested first for a particular SSRC, then
+ * (optionaly) the sync pad can be requested. If no sync pad is requested, no
+ * sync information can be exchanged for this stream.
+ */
+static GstPad *
+gst_rtp_client_request_new_pad (GstElement * element,
+    GstPadTemplate * templ, const gchar * name)
+{
+  GstRTPClient *rtpclient;
+  GstElementClass *klass;
+  GstPadTemplate *rtp_sink_templ, *sync_sink_templ;
+  guint32 ssrc;
+  GstRTPClientStream *stream;
+  GstPad *result;
+
+  g_return_val_if_fail (templ != NULL, NULL);
+  g_return_val_if_fail (GST_IS_RTP_CLIENT (element), NULL);
+
+  if (templ->direction != GST_PAD_SINK)
+    goto wrong_direction;
+
+  rtpclient = GST_RTP_CLIENT (element);
+  klass = GST_ELEMENT_GET_CLASS (element);
+
+  /* figure out the template */
+  rtp_sink_templ = gst_element_class_get_pad_template (klass, "rtp_sink_%d");
+  sync_sink_templ = gst_element_class_get_pad_template (klass, "sync_sink_%d");
+
+  if (templ != rtp_sink_templ && templ != sync_sink_templ)
+    goto wrong_template;
+
+  if (templ == rtp_sink_templ) {
+    /* create new rtp sink pad. If a stream with the pad number already exists
+     * we have an error, else we create the sinkpad, add a jitterbuffer and
+     * ptdemuxer. */
+    if (name == NULL || strlen (name) < 9)
+      goto no_name;
+
+    ssrc = atoi (&name[9]);
+
+    /* see if a stream with that name exists, if so we have an error. */
+    stream = find_stream_by_ssrc (rtpclient, ssrc);
+    if (stream != NULL)
+      goto stream_exists;
+
+    /* ok, create new stream */
+    stream = create_stream (rtpclient, ssrc);
+    if (stream == NULL)
+      goto stream_not_found;
+
+    result = stream->rtp_sink;
+  } else {
+    /* create new rtp sink pad. We can only do this if the RTP pad was
+     * requested before, meaning the session with the padnumber must exist. */
+    if (name == NULL || strlen (name) < 10)
+      goto no_name;
+
+    ssrc = atoi (&name[10]);
+
+    /* find stream */
+    stream = find_stream_by_ssrc (rtpclient, ssrc);
+    if (stream == NULL)
+      goto stream_not_found;
+
+    stream->sync_sink =
+        gst_pad_new_from_static_template (&rtpclient_sync_sink_template, name);
+    gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->sync_sink);
+
+    result = stream->sync_sink;
+  }
+
+  return result;
+
+  /* ERRORS */
+wrong_direction:
+  {
+    g_warning ("rtpclient: request pad that is not a SINK pad");
+    return NULL;
+  }
+wrong_template:
+  {
+    g_warning ("rtpclient: this is not our template");
+    return NULL;
+  }
+no_name:
+  {
+    g_warning ("rtpclient: no padname was specified");
+    return NULL;
+  }
+stream_exists:
+  {
+    g_warning ("rtpclient: stream with SSRC %d already registered", ssrc);
+    return NULL;
+  }
+stream_not_found:
+  {
+    g_warning ("rtpclient: stream with SSRC %d not yet registered", ssrc);
+    return NULL;
+  }
+}
+
+static void
+gst_rtp_client_release_pad (GstElement * element, GstPad * pad)
+{
+}