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, 161 insertions, 0 deletions

diff --git a/gst/rtpmanager/rtpstats.h b/gst/rtpmanager/rtpstats.h
new file mode 100644
index 00000000..66aa7bf7
--- /dev/null
+++ b/gst/rtpmanager/rtpstats.h
@@ -0,0 +1,161 @@
+/* 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.
+ */
+
+#ifndef __RTP_STATS_H__
+#define __RTP_STATS_H__
+
+#include <gst/gst.h>
+#include <gst/netbuffer/gstnetbuffer.h>
+
+/**
+ * RTPSenderReport:
+ *
+ * A sender report structure.
+ */
+typedef struct {
+  gboolean is_valid;
+  guint64 ntptime;
+  guint32 rtptime;
+  guint32 packet_count;
+  guint32 octet_count;
+} RTPSenderReport;
+
+/**
+ * RTPReceiverReport:
+ *
+ * A receiver report structure.
+ */
+typedef struct {
+  gboolean is_valid;
+  guint32 ssrc; /* who the report is from */
+  guint8  fractionlost;
+  guint32 packetslost;
+  guint32 exthighestseq;
+  guint32 jitter;
+  guint32 lsr;
+  guint32 dlsr;
+} RTPReceiverReport;
+
+/**
+ * RTPArrivalStats:
+ * @time: arrival time of a packet
+ * @address: address of the sender of the packet
+ * @bytes: bytes of the packet including lowlevel overhead
+ * @payload_len: bytes of the RTP payload
+ *
+ * Structure holding information about the arrival stats of a packet.
+ */
+typedef struct {
+  GstClockTime  time;
+  gboolean      have_address;
+  GstNetAddress address;
+  guint         bytes;
+  guint         payload_len;
+} RTPArrivalStats;
+
+/**
+ * RTPSourceStats:
+ * @packetsreceived: number of received packets in total
+ * @prevpacketsreceived: number of packets received in previous reporting
+ *                       interval
+ * @octetsreceived: number of payload bytes received
+ * @bytesreceived: number of total bytes received including headers and lower
+ *                 protocol level overhead
+ * @max_seqnr: highest sequence number received
+ * @transit: previous transit time used for calculating @jitter
+ * @jitter: current jitter
+ * @prev_rtptime: previous time when an RTP packet was received
+ * @prev_rtcptime: previous time when an RTCP packet was received
+ * @last_rtptime: time when last RTP packet received
+ * @last_rtcptime: time when last RTCP packet received
+ * @curr_rr: index of current @rr block
+ * @rr: previous and current receiver report block
+ * @curr_sr: index of current @sr block
+ * @sr: previous and current sender report block
+ *
+ * Stats about a source.
+ */
+typedef struct {
+  guint64      packetsreceived;
+  guint64      prevpacketsreceived;
+  guint64      octetsreceived;
+  guint64      bytesreceived;
+  guint16      max_seqnr;
+  guint32      transit;
+  guint32      jitter;
+
+  /* when we received stuff */
+  GstClockTime prev_rtptime;
+  GstClockTime prev_rtcptime;
+  GstClockTime last_rtptime;
+  GstClockTime last_rtcptime;
+
+  /* sender and receiver reports */
+  gint              curr_rr;
+  RTPReceiverReport rr[2];
+  gint              curr_sr;
+  RTPSenderReport   sr[2];
+} RTPSourceStats;
+
+#define RTP_STATS_BANDWIDTH           64000.0
+#define RTP_STATS_RTCP_BANDWIDTH      3000.0
+/*
+ * Minimum average time between RTCP packets from this site (in
+ * seconds).  This time prevents the reports from `clumping' when
+ * sessions are small and the law of large numbers isn't helping
+ * to smooth out the traffic.  It also keeps the report interval
+ * from becoming ridiculously small during transient outages like
+ * a network partition.
+ */
+#define RTP_STATS_MIN_INTERVAL      5.0
+ /*
+ * Fraction of the RTCP bandwidth to be shared among active
+ * senders.  (This fraction was chosen so that in a typical
+ * session with one or two active senders, the computed report
+ * time would be roughly equal to the minimum report time so that
+ * we don't unnecessarily slow down receiver reports.) The
+ * receiver fraction must be 1 - the sender fraction.
+ */
+#define RTP_STATS_SENDER_FRACTION       (0.25)
+#define RTP_STATS_RECEIVER_FRACTION     (1.0 - RTP_STATS_SENDER_FRACTION)
+
+/**
+ * RTPSessionStats:
+ *
+ * Stats kept for a session and used to produce RTCP packet timeouts.
+ */
+typedef struct {
+  gdouble       bandwidth;
+  gdouble       sender_fraction;
+  gdouble       receiver_fraction;
+  gdouble       rtcp_bandwidth;
+  gdouble       min_interval;
+  guint         sender_sources;
+  guint         active_sources;
+  guint         avg_rtcp_packet_size;
+  guint         avg_bye_packet_size;
+  gboolean      sent_rtcp;
+} RTPSessionStats;
+
+void           rtp_stats_init_defaults               (RTPSessionStats *stats);
+
+gdouble        rtp_stats_calculate_rtcp_interval    (RTPSessionStats *stats, gboolean sender);
+gdouble        rtp_stats_add_rtcp_jitter            (RTPSessionStats *stats, gdouble interval);
+
+#endif /* __RTP_STATS_H__ */