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, 162 insertions, 0 deletions

diff --git a/gst/rtpmanager/rtpsource.h b/gst/rtpmanager/rtpsource.h
new file mode 100644
index 00000000..d4ae6f55
--- /dev/null
+++ b/gst/rtpmanager/rtpsource.h
@@ -0,0 +1,162 @@
+/* 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_SOURCE_H__
+#define __RTP_SOURCE_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+#include <gst/netbuffer/gstnetbuffer.h>
+
+#include "rtpstats.h"
+
+/* the default number of consecutive RTP packets we need to receive before the
+ * source is considered valid */
+#define RTP_NO_PROBATION        0
+#define RTP_DEFAULT_PROBATION   2
+
+typedef struct _RTPSource RTPSource;
+typedef struct _RTPSourceClass RTPSourceClass;
+
+#define RTP_TYPE_SOURCE             (rtp_source_get_type())
+#define RTP_SOURCE(src)             (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_SOURCE,RTPSource))
+#define RTP_SOURCE_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SOURCE,RTPSourceClass))
+#define RTP_IS_SOURCE(src)          (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_SOURCE))
+#define RTP_IS_SOURCE_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SOURCE))
+#define RTP_SOURCE_CAST(src)        ((RTPSource *)(src))
+
+/**
+ * RTP_SOURCE_IS_ACTIVE:
+ * @src: an #RTPSource
+ *
+ * Check if @src is active. A source is active when it has been validated
+ * and has not yet received a BYE packet.
+ */
+#define RTP_SOURCE_IS_ACTIVE(src)  (src->validated && !src->received_bye)
+
+/**
+ * RTP_SOURCE_IS_SENDER:
+ * @src: an #RTPSource
+ *
+ * Check if @src is a sender.
+ */
+#define RTP_SOURCE_IS_SENDER(src)  (src->is_sender)
+
+/**
+ * RTPSourcePushRTP:
+ * @src: an #RTPSource
+ * @buffer: the RTP buffer ready for processing
+ * @user_data: user data specified when registering
+ *
+ * This callback will be called when @src has @buffer ready for further
+ * processing.
+ *
+ * Returns: a #GstFlowReturn.
+ */
+typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, gpointer user_data);
+
+/**
+ * RTPSourceClockRate:
+ * @src: an #RTPSource
+ * @payload: a payload type
+ * @user_data: user data specified when registering
+ *
+ * This callback will be called when @src needs the clock-rate of the
+ * @payload.
+ *
+ * Returns: a clock-rate for @payload.
+ */
+typedef gint (*RTPSourceClockRate) (RTPSource *src, guint8 payload, gpointer user_data);
+
+/**
+ * RTPSourceCallbacks:
+ * @push_rtp: a packet becomes available for handling
+ * @clock_rate: a clock-rate is requested
+ * @get_time: the current clock time is requested
+ *
+ * Callbacks performed by #RTPSource when actions need to be performed.
+ */
+typedef struct {
+  RTPSourcePushRTP     push_rtp;
+  RTPSourceClockRate   clock_rate;
+} RTPSourceCallbacks;
+
+/**
+ * RTPSource:
+ *
+ * A source in the #RTPSession
+ */
+struct _RTPSource {
+  GObject       object;
+
+  /*< private >*/
+  RTPSourceCallbacks callbacks;
+  gpointer           user_data;
+
+  guint32       ssrc;
+  gchar        *cname;
+  gint          probation;
+  gboolean      validated;
+  gboolean      received_bye;
+  gchar        *bye_reason;
+
+  gboolean      is_csrc;
+  gboolean      is_sender;
+
+  gboolean      have_rtp_from;
+  GstNetAddress rtp_from;
+  gboolean      have_rtcp_from;
+  GstNetAddress rtcp_from;
+
+  guint8        payload;
+  gint          clock_rate;
+
+  GQueue       *packets;
+
+  RTPSourceStats stats;
+};
+
+struct _RTPSourceClass {
+  GObjectClass   parent_class;
+};
+
+GType rtp_source_get_type (void);
+
+/* managing lifetime of sources */
+RTPSource*      rtp_source_new            (guint32 ssrc);
+
+void            rtp_source_set_callbacks  (RTPSource *src, RTPSourceCallbacks *cb, gpointer data);
+void            rtp_source_set_as_csrc    (RTPSource *src);
+
+void            rtp_source_set_rtp_from   (RTPSource *src, GstNetAddress *address);
+void            rtp_source_set_rtcp_from  (RTPSource *src, GstNetAddress *address);
+
+GstFlowReturn   rtp_source_process_rtp    (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
+
+GstFlowReturn   rtp_source_send_rtp       (RTPSource *src, GstBuffer *buffer);
+
+/* RTCP messages */
+void            rtp_source_process_bye    (RTPSource *src, const gchar *reason);
+void            rtp_source_process_sr     (RTPSource *src, guint64 ntptime, guint32 rtptime,
+                                           guint32 packet_count, guint32 octet_count);
+void            rtp_source_process_rb     (RTPSource *src, guint8 fractionlost, gint32 packetslost,
+                                           guint32 exthighestseq, guint32 jitter,
+                                           guint32 lsr, guint32 dlsr);
+
+#endif /* __RTP_SOURCE_H__ */