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, 350 insertions, 0 deletions

diff --git a/gst/rtpmanager/gstrtpptdemux.c b/gst/rtpmanager/gstrtpptdemux.c
new file mode 100644
index 00000000..5950f619
--- /dev/null
+++ b/gst/rtpmanager/gstrtpptdemux.c
@@ -0,0 +1,350 @@
+/* 
+ * RTP Demux element
+ *
+ * Copyright (C) 2005 Nokia Corporation.
+ * @author Kai Vehmanen <kai.vehmanen@nokia.com>
+ *
+ * Loosely based on GStreamer gstdecodebin
+ * Copyright (C) <2004> 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.
+ */
+
+/*
+ * Contributors:
+ * Andre Moreira Magalhaes <andre.magalhaes@indt.org.br>
+ */
+
+/*
+ * Status:
+ *  - works with the test_rtpdemux.c tool
+ *
+ * Check:
+ *  - is emitting a signal enough, or should we
+ *    use GstEvent to notify downstream elements
+ *    of the new packet... no?
+ *
+ * Notes:
+ *  - emits event both for new PTs, and whenever
+ *    a PT is changed
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <gst/gst.h>
+#include "gstrtpptdemux.h"
+#include <gst/rtp/gstrtpbuffer.h>
+
+/* generic templates */
+static GstStaticPadTemplate rtp_pt_demux_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "payload = (int) [ 0, 255 ], " "clock-rate = (int) [ 0, 2147483647 ]")
+    );
+
+static GstStaticPadTemplate rtp_pt_demux_src_template =
+GST_STATIC_PAD_TEMPLATE ("src%d",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS_ANY);
+
+GST_DEBUG_CATEGORY_STATIC (gst_rtp_pt_demux_debug);
+#define GST_CAT_DEFAULT gst_rtp_pt_demux_debug
+
+/**
+ * Item for storing GstPad<->pt pairs.
+ */
+struct _GstRTPPtDemuxPad
+{
+  GstPad *pad;        /**< pointer to the actual pad */
+  gint pt;             /**< RTP payload-type attached to pad */
+};
+
+/* signals */
+enum
+{
+  SIGNAL_NEW_PAYLOAD_TYPE,
+  SIGNAL_PAYLOAD_TYPE_CHANGE,
+  LAST_SIGNAL
+};
+
+GST_BOILERPLATE (GstRTPPtDemux, gst_rtp_pt_demux, GstElement, GST_TYPE_ELEMENT);
+
+static void gst_rtp_pt_demux_finalize (GObject * object);
+
+static void gst_rtp_pt_demux_release (GstElement * element);
+static gboolean gst_rtp_pt_demux_setup (GstElement * element);
+
+static GstFlowReturn gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf);
+static GstCaps *gst_rtp_pt_demux_getcaps (GstPad * pad);
+static GstStateChangeReturn gst_rtp_pt_demux_change_state (GstElement * element,
+    GstStateChange transition);
+
+static GstPad *find_pad_for_pt (GstRTPPtDemux * rtpdemux, guint8 pt);
+
+static guint gst_rtp_pt_demux_signals[LAST_SIGNAL] = { 0 };
+
+static GstElementDetails gst_rtp_pt_demux_details = {
+  "RTP Demux",
+  /* XXX: what's the correct hierarchy? */
+  "Codec/Demux/Network",
+  "Parses codec streams transmitted in the same RTP session",
+  "Kai Vehmanen <kai.vehmanen@nokia.com>"
+};
+
+static void
+gst_rtp_pt_demux_base_init (gpointer g_class)
+{
+  GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_add_pad_template (gstelement_klass,
+      gst_static_pad_template_get (&rtp_pt_demux_sink_template));
+  gst_element_class_add_pad_template (gstelement_klass,
+      gst_static_pad_template_get (&rtp_pt_demux_src_template));
+
+  gst_element_class_set_details (gstelement_klass, &gst_rtp_pt_demux_details);
+}
+
+static void
+gst_rtp_pt_demux_class_init (GstRTPPtDemuxClass * klass)
+{
+  GObjectClass *gobject_klass;
+  GstElementClass *gstelement_klass;
+
+  gobject_klass = (GObjectClass *) klass;
+  gstelement_klass = (GstElementClass *) klass;
+
+  gst_rtp_pt_demux_signals[SIGNAL_NEW_PAYLOAD_TYPE] =
+      g_signal_new ("new-payload-type",
+      G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTPPtDemuxClass, new_payload_type),
+      NULL, NULL,
+      g_cclosure_marshal_VOID__UINT_POINTER,
+      G_TYPE_NONE, 2, G_TYPE_INT, GST_TYPE_PAD);
+  gst_rtp_pt_demux_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
+      g_signal_new ("payload-type-change",
+      G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTPPtDemuxClass, payload_type_change),
+      NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
+
+  gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_finalize);
+
+  gstelement_klass->change_state =
+      GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_change_state);
+
+  GST_DEBUG_CATEGORY_INIT (gst_rtp_pt_demux_debug,
+      "rtpptdemux", 0, "RTP codec demuxer");
+
+}
+
+static void
+gst_rtp_pt_demux_init (GstRTPPtDemux * ptdemux, GstRTPPtDemuxClass * g_class)
+{
+  GstElementClass *klass = GST_ELEMENT_GET_CLASS (ptdemux);
+
+  ptdemux->sink =
+      gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
+          "sink"), "sink");
+  g_assert (ptdemux->sink != NULL);
+
+  gst_pad_set_chain_function (ptdemux->sink, gst_rtp_pt_demux_chain);
+
+  gst_element_add_pad (GST_ELEMENT (ptdemux), ptdemux->sink);
+}
+
+static void
+gst_rtp_pt_demux_finalize (GObject * object)
+{
+  gst_rtp_pt_demux_release (GST_ELEMENT (object));
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static GstFlowReturn
+gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstRTPPtDemux *rtpdemux;
+  GstElement *element = GST_ELEMENT (GST_OBJECT_PARENT (pad));
+  guint8 pt;
+  GstPad *srcpad;
+
+  rtpdemux = GST_RTP_PT_DEMUX (GST_OBJECT_PARENT (pad));
+
+  g_return_val_if_fail (gst_rtp_buffer_validate (buf), GST_FLOW_ERROR);
+
+  pt = gst_rtp_buffer_get_payload_type (buf);
+
+  srcpad = find_pad_for_pt (rtpdemux, pt);
+  if (srcpad == NULL) {
+    /* new PT, create a src pad */
+    GstElementClass *klass;
+    GstPadTemplate *templ;
+    gchar *padname;
+    GstCaps *caps;
+    GstRTPPtDemuxPad *rtpdemuxpad;
+
+    klass = GST_ELEMENT_GET_CLASS (rtpdemux);
+    templ = gst_element_class_get_pad_template (klass, "src%d");
+    padname = g_strdup_printf ("src%d", pt);
+    srcpad = gst_pad_new_from_template (templ, padname);
+    g_free (padname);
+
+    caps = gst_pad_get_caps (srcpad);
+    caps = gst_caps_make_writable (caps);
+    gst_caps_append_structure (caps,
+        gst_structure_new ("payload", "payload", G_TYPE_INT, pt, NULL));
+    gst_pad_set_caps (srcpad, caps);
+
+    /* XXX: set _link () function */
+    gst_pad_set_getcaps_function (srcpad, gst_rtp_pt_demux_getcaps);
+    gst_pad_set_active (srcpad, TRUE);
+    gst_element_add_pad (element, srcpad);
+
+    if (srcpad) {
+      GST_DEBUG ("Adding pt=%d to the list.", pt);
+      rtpdemuxpad = g_new0 (GstRTPPtDemuxPad, 1);
+      rtpdemuxpad->pt = pt;
+      rtpdemuxpad->pad = srcpad;
+      rtpdemux->srcpads = g_slist_append (rtpdemux->srcpads, rtpdemuxpad);
+
+      GST_DEBUG ("emitting new-payload_type for pt %d", pt);
+      g_signal_emit (G_OBJECT (rtpdemux),
+          gst_rtp_pt_demux_signals[SIGNAL_NEW_PAYLOAD_TYPE], 0, pt, srcpad);
+    }
+  }
+
+  if (pt != rtpdemux->last_pt) {
+    gint emit_pt = pt;
+
+    /* our own signal with an extra flag that this is the only pad */
+    rtpdemux->last_pt = pt;
+    GST_DEBUG ("emitting payload-type-changed for pt %d", emit_pt);
+    g_signal_emit (G_OBJECT (rtpdemux),
+        gst_rtp_pt_demux_signals[SIGNAL_PAYLOAD_TYPE_CHANGE], 0, emit_pt);
+  }
+
+  /* push to srcpad */
+  if (srcpad)
+    gst_pad_push (srcpad, GST_BUFFER (buf));
+
+  return ret;
+}
+
+static GstCaps *
+gst_rtp_pt_demux_getcaps (GstPad * pad)
+{
+  GstCaps *caps;
+
+  GST_OBJECT_LOCK (pad);
+  if ((caps = GST_PAD_CAPS (pad)))
+    caps = gst_caps_ref (caps);
+  GST_OBJECT_UNLOCK (pad);
+
+  return caps;
+}
+
+static GstPad *
+find_pad_for_pt (GstRTPPtDemux * rtpdemux, guint8 pt)
+{
+  GstPad *respad = NULL;
+  GSList *item = rtpdemux->srcpads;
+
+  for (; item; item = g_slist_next (item)) {
+    GstRTPPtDemuxPad *pad = item->data;
+
+    if (pad->pt == pt) {
+      respad = pad->pad;
+      break;
+    }
+  }
+
+  return respad;
+}
+
+/**
+ * Reserves resources for the object.
+ */
+static gboolean
+gst_rtp_pt_demux_setup (GstElement * element)
+{
+  GstRTPPtDemux *ptdemux = GST_RTP_PT_DEMUX (element);
+  gboolean res = TRUE;
+
+  if (ptdemux) {
+    ptdemux->srcpads = NULL;
+    ptdemux->last_pt = 0xFFFF;
+  }
+
+  return res;
+}
+
+/**
+ * Free resources for the object.
+ */
+static void
+gst_rtp_pt_demux_release (GstElement * element)
+{
+  GstRTPPtDemux *ptdemux = GST_RTP_PT_DEMUX (element);
+
+  if (ptdemux) {
+    /* note: GstElement's dispose() will handle the pads */
+    g_slist_free (ptdemux->srcpads);
+    ptdemux->srcpads = NULL;
+  }
+}
+
+static GstStateChangeReturn
+gst_rtp_pt_demux_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret;
+  GstRTPPtDemux *ptdemux;
+
+  ptdemux = GST_RTP_PT_DEMUX (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (gst_rtp_pt_demux_setup (element) != TRUE)
+        ret = GST_STATE_CHANGE_FAILURE;
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      gst_rtp_pt_demux_release (element);
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}