configure.ac: Disable rtpmanager for now because it depends on CVS -base.

Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.

1 files changed, 111 insertions, 0 deletions

diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c
new file mode 100644
index 00000000..b9076eac
--- /dev/null
+++ b/gst/rtpmanager/rtpstats.c
@@ -0,0 +1,111 @@
+/* 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.
+ */
+
+#include "rtpstats.h"
+
+/**
+ * rtp_stats_init_defaults:
+ * @stats: an #RTPSessionStats struct
+ *
+ * Initialize @stats with its default values.
+ */
+void
+rtp_stats_init_defaults (RTPSessionStats * stats)
+{
+  stats->bandwidth = RTP_STATS_BANDWIDTH;
+  stats->sender_fraction = RTP_STATS_SENDER_FRACTION;
+  stats->receiver_fraction = RTP_STATS_RECEIVER_FRACTION;
+  stats->rtcp_bandwidth = RTP_STATS_RTCP_BANDWIDTH;
+  stats->min_interval = RTP_STATS_MIN_INTERVAL;
+}
+
+/**
+ * rtp_stats_calculate_rtcp_interval:
+ * @stats: an #RTPSessionStats struct
+ * 
+ * Calculate the RTCP interval. The result of this function is the amount of
+ * time to wait (in seconds) before sender a new RTCP message.
+ *
+ * Returns: the RTCP interval.
+ */
+gdouble
+rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean sender)
+{
+  gdouble active, senders, receivers, sfraction;
+  gboolean avg_rtcp;
+  gdouble interval;
+
+  active = stats->active_sources;
+  /* Try to avoid division by zero */
+  if (stats->active_sources == 0)
+    active += 1.0;
+
+  senders = (gdouble) stats->sender_sources;
+  receivers = (gdouble) (active - senders);
+  avg_rtcp = (gdouble) stats->avg_rtcp_packet_size;
+
+  sfraction = senders / active;
+
+  GST_DEBUG ("senders: %f, receivers %f, avg_rtcp %f, sfraction %f",
+      senders, receivers, avg_rtcp, sfraction);
+
+  if (sfraction <= stats->sender_fraction) {
+    if (sender) {
+      interval =
+          (avg_rtcp * senders) / (stats->sender_fraction *
+          stats->rtcp_bandwidth);
+    } else {
+      interval =
+          (avg_rtcp * receivers) / ((1.0 -
+              stats->sender_fraction) * stats->rtcp_bandwidth);
+    }
+  } else {
+    interval = (avg_rtcp * active) / stats->rtcp_bandwidth;
+  }
+
+  if (interval < stats->min_interval)
+    interval = stats->min_interval;
+
+  if (!stats->sent_rtcp)
+    interval /= 2.0;
+
+  return interval;
+}
+
+/**
+ * rtp_stats_calculate_rtcp_interval:
+ * @stats: an #RTPSessionStats struct
+ * @interval: an RTCP interval
+ * 
+ * Apply a random jitter to the @interval. @interval is typically obtained with
+ * rtp_stats_calculate_rtcp_interval().
+ *
+ * Returns: the new RTCP interval.
+ */
+gdouble
+rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, gdouble interval)
+{
+  /* see RFC 3550 p 30 
+   * To compensate for "unconditional reconsideration" converging to a
+   * value below the intended average.
+   */
+#define COMPENSATION  (2.71828 - 1.5);
+
+  return (interval * g_random_double_range (0.5, 1.5)) / COMPENSATION;
+}